SCIENTIFIC AND TECHNICAL AEROSPACE REPORTS
A Biweekly Publication of the National Aeronautics and Space Administration
VOLUME 44, ISSUE 6 - March 24, 2006
33 ELECTRONICS AND ELECTRICAL ENGINEERING
Includes development, performance, and maintainability of electrical/electronic devices and components; related test equipment; and microelectronics and integrated circuitry.
For related information see also 60 Computer Operations and Hardware; and 76 Solid-State Physics.
For communications equipment and devices see 32 Communications and Radar.
20060007991 Pacific Northwest National Lab., Richland, WA, USA
FY 2005 Infrared Photonics
Anheier, N. C.; Allen, P. J.; Keller, P. E.; Bennett, W. D.; Martin, P. M.; Oct. 2004; 52 pp.; In English Report No.(s): DE2005-15020768; PNNL-15209; No Copyright; Avail.: National Technical Information Service (NTIS)
Research done by the Infrared Photonics team at Pacific Northwest National Laboratory (PNNL) is focused on developing miniaturized integrated optics for mid-wave infrared (MWIR) and long-wave infrared (LWIR) sensing applications by exploiting the unique optical and material properties of chalcogenide glass. PNNL has developed thin-film deposition capabilities, direct laser writing techniques, infrared photonic device demonstration, holographic optical element design and fabrication, photonic device modeling, and advanced optical metrology--all specific to chalcogenide glass. Chalcogenide infrared photonics provides a pathway to quantum cascade laser (QCL) transmitter miniaturization. QCLs provide a viable infrared laser source for a new class of laser transmitters capable of meeting the performance requirements for a variety of national security sensing applications. The high output power, small size, and superb stability and modulation characteristics of QCLs make them amenable for integration as transmitters into ultra-sensitive, ultra-selective point sampling and remote short-range chemical sensors that are particularly useful for nuclear nonproliferation missions. NTIS
Chalcogenides; Infrared Radiation; Photonics
20060008079 Bureau of the Census, Washington, DC, USA
Economic Census 2002: Wholesale Trade, Industry Series. Electrical, Hardware, Plumbing, and Heating Equipment and Supplies
Nov. 2004; 68 pp.; In English Report No.(s): PB2006-103798; EC02-42I-17; No Copyright; Avail.: CASI: A04, Hardcopy
The economic census is the major source of facts about the structure and functioning of the nations economy. It provides essential information for government, business, industry, and the general public. Title 13 of the USA Code (Sections 131, 191, and 224) directs the Census Bureau to take the economic census every 5 years, covering years ending in 2 and 7. The economic census furnishes an important part of the framework for such composite measures as the gross domestic product estimates, input/output measures, production and price indexes, and other statistical series that measure short-term changes in economic conditions. NTIS
Census; Classifications; Economics; Electric Equipment; Fluid Flow; Heating; Heating Equipment; Pipelines; Supplying
20060008083 Bureau of the Census, Washington, DC, USA
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Economic Census 2002: Wholesale Trade, Industry Series. Electronic Markets and Agents and Brokers
Oct. 2004; 62 pp.; In English Report No.(s): PB2006-103800; EC02-42I-19; No Copyright; Avail.: CASI: A04, Hardcopy
The economic census is the major source of facts about the structure and functioning of the nations economy.
It provides essential information for government, business, industry, and the general public.
Title 13 of the USA Code (Sections 131, 191, and 224) directs the Census Bureau to take the economic census every 5 years, covering years ending in 2 and 7.
The economic census furnishes an important part of the framework for such composite measures as the gross domestic product estimates, input/output measures, production and price indexes, and other statistical series that measure short-term changes in economic conditions. NTIS
Census; Classifications; Economic Analysis; Economics; Market Research
20060008117 Tsinghua Univ., Bejing, China
Development of a Portable Two-channel EEG Radio Telemetry System
Jiang, Xin; Wang, Xiao-guang; Space Medicine and Medical Engineering, Volume 18, No. 6; December 2005, pp. 451-455; In Chinese; See also 20060008103; Copyright; Avail.: Other Sources
Objective: To develop a set of portable two-channel EEG radio-telemetry system. Method: 1) EEG amplifier: using instrument amplifier AD8221 for preamplifier and precision amplifier AD824 for filtering, amplifying and voltage adjustment; 2) Wireless transmission: this part was based on nRF24E1 that is a 2.4 GHz transceiver with embedded 8051 core and 8 input ADC. Software programming was used to realize two-channel data acquisition and encapsulation; 3) Power supply: Li battery was used to provide charging and discharging management and alarm function. Result: It was proved experimentally that this system could meet the requirement of a portable EEG acquisition system. Conclusion: The design of the portable EEG radio telemetry system is practical and can realize headset system after some improvements. Author
Electroencephalography; Radio Telemetry; Preamplifiers; Electric Batteries; Amplification; Transmitter Receivers; Warning Systems
20060008123 Rostock Univ., Germany
Transmit Power Allocation for BER Performance Improvements in SVD equalized Wavelet-based Multicarrier Systems
Ahrens, Andreas; Lange, Christoph; International Conference on Advances in the Internet, Processing, Systems and Interdisciplinary Research (IPSI-2005 FRANCE); [2005]; 5 pp.; In English; See also 20060008121; Original contains black and white illustrations; Copyright; Avail.: CASI: A01, Hardcopy; Available from CASI on CD-ROM only as part of the entire parent document
In this contribution transmit power allocation schemes for BER performance improvements in a Singular Value Decomposition (SVD) equalized wavelet-based multicarrier system are investigated under the constraint of limited total transmit power. Contrary to other contributions we don t adapt the transmit power to the subchannels. Here, we adapt the transmit power based on a SVD equalization strategy directly to the data symbol amplitudes of the transmit vector. We investigate different optimal and suboptimal solutions based on the Lagrange multiplier method. Author
Lagrange Multipliers; Bit Error Rate; Wavelet Analysis
20060008153 Ottawa Univ., Ontario, Canada
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Digital Signal Processing for Tracking in Wireless Sensor Networks
Bolic, Miodrag; International Conference on Advance in the Internet, Processing, Systems and Interdisciplinary Research (IPSI-2005 BELGRADE); [2005]; 4 pp.; In English; See also 20060008152; Original contains black and white illustrations; Copyright; Avail.: CASI: A01, Hardcopy; Available from CASI on CD-ROM only as part of the entire parent document
The field of wireless sensor networks is emerging and attracts a lot of attention in the computer science and electrical engineering research communities. In this paper, we study the characteristics of statistical signal processing algorithms called particle filters used for tracking in wireless sensor networks. These algorithms are usually too computationally complex and do not go together with the low power requirements of wireless sensor network nodes. We show several types of implementation of sensor nodes and analyze which architectures will be suitable for implementation of particle filters. We propose some direction on how to decrease the complexity of the algorithms as well as how to reduce communication requirements caused by particle filtering. Author
Signal Processing; Tracking Networks; Electrical Engineering; Sensors; Wireless Communication
20060008156 Skvode Univ., Skvode, Sweden
Smart Antennas in Mobile Computing
ShanmukhaShri, Varada Sri; International Conference on Advance in the Internet, Processing, Systems and Interdisciplinary Research (IPSI-2005 BELGRADE); [2005]; 45 pp.; In English; See also 20060008152; Original contains color and black and white illustrations; Copyright; Avail.: CASI: A03, Hardcopy; Available from CASI on CD-ROM only as part of the entire parent document
The concept of using multiple antennas and innovative signal processing to serve cells more intelligently has existed for many years. In fact, varying degrees of relatively costly smart antenna systems have already been applied in defense systems. Until recent year, cost barrier have prevented their use in commercial systems. The advent of powerful low-cost digital signal processors (DSPs), general purpose processors and ASICs, as well as innovative software-based signal processing techniques (algorithms) have made intelligent antennas practical for cellular communications systems or in other terms mobile computing. Derived from text
Signal Processing; Signal Analyzers; Telecommunication; Antenna Design
20060008688 Illinois Inst. of Tech., Chicago, IL, USA
Electromagnetic and Mechanical Design of Gridded Radio- Frequency Cavity Windows
Alsharo'a, M. M.; Dec. 2004; 96 pp.; In English Report No.(s): DE2005-15020218; No Copyright; Avail.: National Technical Information Service (NTIS)
No abstract available
Cavities; Mechanical Engineering; Radio Frequencies
20060008750 Rensselaer Polytechnic Inst., Troy, NY, USA
Image Restoration by the Method of Gneralized Projections with Application to Restoration from Magnitude
Levi, Aharon; Stark, Henry; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.5.1 - 37.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Image restoration problems which involve nonconvex constrains such as the restoration from magnitude problem (RFM) cannot be solved by the method of projections onto convex sets (POCS). An algorithm known as generalized projections is derived and its properties are discussed. We introduce a performance criterion which can be used to measure the performance of the algorithm at any iteration. A method of optimizing (via control of relaxation parameters) the algorithm is described. Comparisons with other RFM methods are furnished. Author
Restoration; Iteration; Imagery; Set Theory
20060008751 Arizona Univ., Tucson, AZ, USA
EXPERIMENTS ON the USE of LOCAL STATISTICS FOR ADAPTIVE IMAGE PROCESSING
Strickland, Robin N.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.7.1 - 37.7.4; In English; See also 20060008748 Contract(s)/Grant(s): AFOSR-81-0170; Copyright; Avail.: Other Sources
In this paper, we report results from our experiments to find useful measures of local image statistics from small sub-blocks of data. Our criteria for the reliability of statistical estimates is that they should yield high quality results when used in adaptive image processing. We describe a method for estimating the correlation parameters of first-order Markov auto-covariance models using relatively few computational steps. An application to block-interpolative data compression is given Author
Covariance; Image Processing; Data Compression; Estimates
20060008752 National Univ. of Singapore, Singapore
A TECHNIQUE FOR SPECTRAL COMPONENT LOCATION WITHIN A FFT RESOLUTION CELL
Ng, S. S.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.8.1 - 38.8.3; In English; See also 20060008748; Copyright; Avail.: Other Sources
A technique has been developed to locate the peak of a spectral component within a resolution cell of a Fast Fourier Transform Spectrum. An iterative approximate curve fitting process is applied to the spectral magnitude at two adjacent frequency points in the immediate vicinity of the expected component peak. A correction factor is computed from which the exact location of the component peak can be determined. Author
Fast Fourier Transformations; Resolution Cell; Iteration; Curve Fitting; Frequencies
20060008753 Massachusetts Inst. of Tech., Cambridge, MA, USA
Knowledge Based Speech Analysis and Enhancement
IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 39A.4.1 - 39A.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper describes a system for speech analysis and enhancement which combines signal processing and symbolic processing in a closely coupled manner. The system takes as input both a noisy speech signal and a symbolic description of the speech signal. The system attempts to reconstruct the original speech waveform using symbolic processing to help model the signal and to guide reconstruction. The system uses various signal processing algorithms for parameter estimation and reconstruction. Author
Knowledge Based Systems; Speech Recognition; Parameter Identification; Signal Processing; Systems Analysis
20060008756 Philips Research Labs., Eindhoven, Netherlands
GABOR REPRESENTATION and WIGNER DISTRIBUTION of SIGNALS
Janssen, A. J. E. M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.2.1 - 41B.2.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
We compare Gabor representation and the Wigner distribution on their merits for the time-frequency description of signals. Author
Time Signals; Frequencies; Quantum Mechanics; Wigner Coefficient; Gabor Transformation; Signal Analysis
20060008759 Oki Electric Industry Ltd., Tokyo, Japan
A DIGITAL SIGNAL PROCESSOR MODULE ARCHITECTURE and ITS IMPLEMENTATION USING VLSIS
Okada, K.; Ehara, T.; Suzuki, H.; Yanagida, K.; Saito, K.; Ichiura, N.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.5.1 - 44.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A digital signal processor module (DSPM) has been developed for use in real-time processing of signals from acoustic arrays. This newly developed DSPM is characterized by flexibility; it is compact and gives high performance, and can be used in a wide range of applications. To make this new module, three new VLSIs for digital signal processing were developed: a digital signal processor, an address generator and an input/output port. This new DSPM has a processing capacity of 5 MIPS. The circuitry is fabricated on two printed circuit boards measuring 297 mm x 210 mm. Author
Digital Systems; Signal Processing; Very Large Scale Integration; Acoustics; Fabrication; Architecture (Computers); Electronic Modules
20060008760 Philips G.m.b.H., Hamburg, Germany
A DFT-based Front-End for Word Recognition Systems
Geppert, R.; Schwartau, P.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.14.1 - 44.14.4; In English; See also 20060008748 Contract(s)/Grant(s): BMFT-IT-1501-6; Copyright; Avail.: Other Sources
Combining a custom LSI chip-set for OFT computation and user-programmable signal processors a versatile front-end for word recognition systems has been set up. Depending on the complexity of the required processing algorithms up to four signal processors can be employed in a pipelined architecture. The Multibus-compatible board performs functions like spectral analysis, spectral smoothing, intensity normalisation, and filter bank realization on speech signals sampled at rates of up to 8 kHz. Author
Chips (Electronics); Spectrum Analysis; Words (Language); Signal Processing; Signal Analyzers; Algorithms; Pipelining (Computers)
20060008761 Tektronix, Inc., Beaverton, OR, USA
On All-Zero Modelling of a Recursive Digital Filter
Lo, Pei-hwa; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.2.1 - 45.2.2; In English; See also 20060008748; Copyright; Avail.: Other Sources
In this paper, we investigate the feasibility and limitations of modelling a recursive digital system using an all-zero digital filter. It is shown that the accuracy of modelling depends on the pole of the recursive system and the number of collected data points. It is also shown that the long division method can determine the optimal performance for any given number of FIR digital filter stages. Author
Digital Filters; Fir Filters; IIR Filters; Digital Systems
20060008762 Academia Sinica, Beijing, China
THE GENERALIZED PHASE SPECTRUM METHOD FOR TIME DELAY ESTIMATION
Zhen, Zhao; Zi-qiang, Hou; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.2.1 - 46.2.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The concept of the Generalized Phase Spectrum (GPS) TDE is put forward. The relation between the GPS TDE and the GCC TDE is derived. A multipath signal model is considered. The method of Amplitude Square (AS) weighting for TDE is proposed and the comparison between the AS TDE and the Phase Data (PD) TDE is made in the multipath environment. The results of theoretical calculation and computer simulation experiment show that the performance of the AS weighting TDE is superior to that of the PD TDE. Author
Time Lag; Computerized Simulation; Multipath Transmission
20060008766 Rhode Island Univ., Kingston, RI, USA
Accuracy of Frequency Estimation and Its Relation to Prediction Filter Order
Tufts, D. W.; Kumaresan, R.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.9.1 - 38.9.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-81-K-0144; Copyright; Avail.: Other Sources
Estimates of frequencies of noise corrupted sinusoidal signals are often obtained from the zero locations of data adaptive or prediction error filters. It is well known that the frequency estimation error decreases as the order of the prediction error filter is increased for a given data record length. In this paper, we explain this phenomenon, at high SNR by calculating the statistics of the estimation error, Author
Accuracy; Linear Prediction; Frequencies; Adaptive Filters
20060008771 Massachusetts Inst. of Tech., Cambridge, MA, USA
ONE-DIMENSIONAL PROCESSING FOR ADAPTIVE IMAGE RESTORATION
Chan, Philip; Lim, Jae S.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.3.1 - 37.3.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-81-K-0742; NR-049-506; ECS80-07102; Copyright; Avail.: Other Sources
In this paper, we present a one-dimensional (1-D) approach to the problem of image restoration. Our approach involves a cascade of four 1-D adaptive filters oriented in the four major correlation directions of the image, with each filter treating the image as a 1-D signal. The objective of our 1-D approach is to improve the performance of the more general two-dimensional (2-D) approach. This differs considerably from previous 1-D approaches, the objectives of which have typically been to approximate a more general 2-D approach for computational reasons and not to improve its performance. To illustrate this point, our approach is applied to an existing 2-D image restoration algorithm. Experiments with images at low SNRs (signal to noise ratios) show that the 1-D approach performs better than the 2-D approach for the specific image restoration algorithm. Our 1-D approach preserves edges while removing noise in all regions of the image including the edge regions. Author
Image Processing; Restoration; Algorithms; Adaptive Filters; Signal to Noise Ratios
20060008772 Massachusetts Inst. of Tech., Lexington, MA, USA
HOMOMORPHIC RESTORATION of IMAGES DEGRADED BY LIGHT CLOUD COVER
Peli, T.; Quatieri, T. F.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.8.1 - 37.8.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
In this paper, we demonstrate the use of adaptive homomorphic filtering in exposing objects under light cloud cover. In particular, the homomorphic filter invoked is space-varying and is parameterized by the local mean level of the degraded image. The local mean serves as an indication of the extent of local cloud cover degradation. We show that this adaptive procedure has greater potential than the long-space methods in exposing objects beneath light cloud cover. In addition, adaptive homomorphic filtering compares favorably with an iterative homomorphic enhancement procedure which is an extension of the one-pass nonadaptive homomorphic filter. Author
Homomorphisms; Adaptive Filters; Restoration; Iteration; Cloud Cover
20060008776 Chalmers Univ. of Technology, Goeteborg, Sweden
COMPUTATIONALLY EFFICIENT ESTIMATION of THE MEAN FREQUENCY FOR REAL-VALUED SIGNALS
Sjoeberg, Sten; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.3.1 - 38.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
It can be shown that the mean frequency of a real-valued stochastic signal can be expressed as an integral of the normalized autocorrelation function r(Tau) weighted by a function equal to 1/Tau(sup 2). The fast decline of the weighting function implies that the behavior of the autocorrelation function for small values of T is the most important portion for estimation of the mean frequency of a signal. It is demonstrated in a simulation study that estimates of the mean frequency with mean squared error equal to the error in estimates obtained via a FFT derived mean frequency estimate can be obtained by using just a few lags of the normalized autocorrelation function with a computational effort substantially less than that required for estimation via FFT. Upper bounds, that can be used as guidelines when implementing the estimator, are given for the bias error introduced by using just a few lag values of the autocorrelation function. Author
Autocorrelation; Mean Square Values; Fast Fourier Transformations; Error Analysis; Estimates
20060008777 Rome Univ., Rome, Italy
EXPLICIT FORMULAS FOR ESTIMATING the FREQUENCIES of SINE WAVES IN NOISE WITH FEW SAMPLES
Martinelli, G.; Orlandi, G.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.5.1 - 38.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Explicit formulas for estimating the frequency of a sinusoid in white noise are proposed. The accuracy of the estimate obtained by these formulas is very close to the Cramer-Rao bound in the case of very short sequences. Author
Cramer-Rao Bounds; Frequencies; Sine Waves; White Noise; Accuracy
20060008780 Forschungsinstitut fuer Hochfrequenzphysik, Werthhoven, Germany
A POINT MECHANICAL MODEL FOR the DYNAMICS of TOWED ARRAYS
Brandenbug,W.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.3.1 - 4.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A point mechanical model for towed arrays is developed. The equations of motion are calculated using the Lagrange formalism of classical mechanics. There are a lot of free parameters in the model, which is demonstrated in two examples. One example shows the bearing and depth of some points of the towed arrays as a function of time while the ship changes the course. In another example an optimal course is shown, which minimizes the curvature of the array. Author
Time Dependence; Classical Mechanics; Curvature; Equations of Motion; Ships
20060008782 Tohoku Univ., Sendai, Japan
COMPOSITE COMPLEX SINUSOIDAL MODELING FOR the ESTIMATION of DIRECTIONS and SPECTRA of INCIDENT PLANE WAVES
Abe, Masato; Kido, Keniti; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.9.1 - 40.9.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
We have already reported that composite complex sinusoidal modeling (CCSM) is effective for array signal processing, in the sense that the highest bearing resolution was achieved by the CCSM than by any other existing methods at that time. The directivity pattern obtained by CCSM has no side lobe and gives only lines indicating the directions and the powers of incident plane waves at a concerning frequency.We propose here an improved CCSM to achieve a much more precise bearing estimation. Author
Signal Processing; Frequencies; Plane Waves; Sine Waves
20060008783 Imperial Coll. of Science and Technology, London, UK
HARDWARE REALIZATION of MERSENNE NUMBER TRANSFORMS FOR FAST DIGITAL CONVOLUTION
Siu, Wan-Chi; Constantinides, A. G.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41A.2.1 - 41A.2.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
In this paper we convert the Mersenne-prime and Mersenne-composite Number Transforms into recursive filter form and propose simple hardware structures to carry out the fast implementation of circular convolutions. We also present the results of our study employing efficient methods to compute long circular convolutions using the Mersenne Number Transforms. Author
IIR Filters; Transformations (Mathematics); Computation
20060008784 National Science Foundation, Arlington, VA, USA
A RESIDUE to MIXED RADIX CONVERTER and ERROR CHECKER FOR A FIVE-MODULI RESIDUE NUMBER SYSTEM
Bell, M. J., Jr.; Jenkins, W. K.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41A4.1 - 41A4.4; In English; See also 20060008748 Contract(s)/Grant(s): NSF ENG-79-01686; Copyright; Avail.: Other Sources
A design is presented for an experimental device which converts data from residue representation to mixed radix representation while simultaneously checking for single digit errors. The experimental system has a high speed pipelined architecture and operates with five 5-bit moduli, two of which are redundant. Experience gained through experimentation with this system led to the design of a universal building block that is proposed for monolithic fabrication. Author
Residues; Errors; Digital Systems; Systems Engineering; Signal Processing
20060008785 Hunter Coll., New York, NY, USA
DISTRIBUTIONS IN SIGNAL THEORY
Cohen, Leon; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.1.1 - 41B.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The general equation for joint time-frequency distribution functions is reformulated to make it more tractible for the study of the recently found positive distributions. Author
Frequency Distribution; Distribution Functions; Temporal Distribution
20060008786 Laboratoire Traitement du Signal et Instrumentation, Saint-Etienne, France
SOME FEATURES of TIME-FREQUENCY REPRESENTATIONS of MULTICOMPONENT SIGNALS
Flandrin, Patrick; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.4.1 - 41B.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The Wigner-Ville Distribution (WVD) is now known to he a convenient tool for the time-frequency ana]ysis of non-stationary signals, and especially monocomponent ones. However, in the case of multicomponent signals, its bilinear structure is also known to create cross-terms without any physical significance. Starting with the general formulation of time-frequency representations, which only depend on an arbitrary kernel function, we first characterize properties of such cross-terms and then propose appropriate smoothings of the WVD in order to reduce their influence. Such suitable and versatile approximations are compared on synthetic and natural signals and an extension to time-frequency filtering is proposed. Author
Kernel Functions; Time Signals; Signal Analysis; Quantum Mechanics
20060008787 Breed and Harvel Associates, Austin, TX, USA
A RANGE and AZIMUTH ESTIMATOR BASED ON FORMING the SPATIAL WIGNER DISTRIBUTION
Breed, B. R.; Posch, Theodore E.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.9.1 - 41B.9.2; In English; See also 20060008748; Copyright; Avail.: Other Sources
We show that for a linear, equally spaced array, wavefront curvature produces a quadratic phase variation across the array for a single arrival frequency from a target within the Fresnel zone of the array. The Wigner distribution for a quadratic phase signal peaks at the instantaneous frequency or, in our case, at the wave number versus array position. By relating this to wavefront curvature, a range estimate is produced. Author
Linear Arrays; Spatial Distribution; Wave Fronts; Quantum Mechanics; Wigner Coefficient
20060008795 Waterloo Univ., Ontario, Canada
AN IEEE696 COMPATIBLE SIGNAL PROCESSOR
Justice, G. A. C.; Mavaddat, F.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.6.1 - 44.6.4; In English; See also 20060008748 Contract(s)/Grant(s): NSERC-A5515; Copyright; Avail.: Other Sources
A general purpose signal processor was designed as a low-cost subsystem for a microcomputer system. This approach allows considerable flexibility in programming the device with only a slight sacrifice in computational throughput. Firmware defines the kernel operations of the signal processor, and allows for the downloading of application code from a host. A speech synthesizer has been implemented on the signal processor and is described as an example of the development flexibility obtained with this system. This signal processor can be placed in any IEEE696 based computer system and the user will be able to utilize the facilities of that system in developing signal processing software. Author
Microcomputers; Signal Processing; Systems Compatibility; Systems Engineering
20060008796 Rockwell International Corp., USA
ARCHITECTURE and CONTROL of A DISTRIBUTED SIGNAL PROCESSOR
Ashcraft, W. Don; South, Hughe M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.8.1 - 44.8.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The SPAN-I is a distributed, digital signal processing system which can perform real-time operations such as spectral analysis at input data rates up to 40 Mbits/sec. The processor consists of twelve major components connected to a ring bus operating at 64 Mbytes/sec. In addition to the transfer of both instructions and data between units, the bus supports 'party lines' which allow data to be sent to multiple destinations. A unique software structure, supported by the bus hardware, achieves the synchronous parallel operation of all units. Author
Digital Systems; Signal Processing; Architecture (Computers); Control Systems Design; Distributed Processing
20060008797 Communications Research Centre, Ottawa, Ontario, Canada
Implementation of a Full Duplex 2.4 kbps LPC Vocoder on a Single TMS-320 Microprocessor Chip
Bryden, Brian; Hassanein, Hisham; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.12.1 - 44.12.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
With the commercial availability of high speed digital signal processors, it is now possible to implement all the linear predictive coding (LPC) tasks (excluding D-A/A-D conversion) on a single chip. In this paper, a very small, high quality, full-duplex, 10th order 2.4 kbps LPC vocoder is described. A single Texas Instruments TMS-320 microprocessor performs LPC analysis, pitch detection, synthesis, and data I/O. At the time of writing this paper, a total of 20 off-the-shelf integrated circuits were used occupying two thirds of a 14cm x 18cm wirewrap board (excluding power supply). The total power dissipation is less than 2 watts. The chip count may be reduced by a factor of two by combining the random logic on a semi-custom integrated circuit. When produced commercially, the cost of this vocoder should be considerably less than existing LPC units. Author
Microprocessors; Signal Analyzers; Vocoders; Integrated Circuits; Chips; Coding; Signal Processing
20060008798 Massachusetts Inst. of Tech., Lexington, MA, USA
A CCD Chip for Parallel Pulse-Doppler Radar Processing
Chiang, A. M.; Shaw, G. A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.15.1 - 44.15.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The generic signal processing requirements for a pulse-Doppler radar are presented and a CCD device for performing a portion of the signal processing is described. Author
Charge Coupled Devices; Parallel Processing (Computers); Pulse Doppler Radar; Signal Processing; Chips (Electronics); Fabrication
20060008799 University of the Budeswehr-Munich, France
SYSTEM IDENTIFICATION TECHNIQUES FOR NOISE REDUCTION IN EVOKED POTENTIAL PROCESSING
Rauner, H.; Wolf, W.; Appel, U.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.1.1 - 45.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Transient visual evoked potentials (EPs) are very small variations of the electroencephalogram (EEG) in response to the application of light stimuli. Because of their small amplitudes compared to those of the on-going EEG, signal extraction methods are necessary to estimate their waveforms. In order to develop improved estimation schemes an engineering model is proposed that accounts for most of the observed effects. This model includes the consideration of the information available from the EEG preceding every onset of a sensory stimulus as basis of signal enhancement by adaptive processing. System identification techniques can be applied to test the model structure and to specify its parameters. Several approaches and first results are discussed. Author
Noise Reduction; System Identification; Parameterization; Evoked Response (Psychophysiology)
20060008800 Strathclyde Univ., Glasgow, UK
ASYMPTOTIC PERFORMANCE of EIGENSTRUCTURE SPECTRAL ANALYSIS METHODS
Sharman, K.; Durrani, T. S.;Wax, M.; Kailath, T.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.5.1 - 45.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper considers some asymptotic statistical properties of covariance eigenstructure spectral analysis techniques. It is shown that when the signal model is of the appropriate form, and the observations are Gaussian, the signal parameter estimates, obtained by locating the nulls in the eigen-spectrum, are asymptotically zero mean normal random variables. Based on this observation, the paper then considers the formation of confidence regions for the signal parameters. The paper presents the general case of a multi-dimensional eigenstructure algorithm, which estimates one or more parameters of each signal in the observed data. Author
Spectrum Analysis; Asymptotic Properties; Parameter Identification; Statistical Distributions; Covariance; Random Variables
20060008801 Arizona State Univ., Tempe, AZ, USA
ARMA SYSTEM IDENTIFICATION: AN ALGEBRAIC APPROACH
Cadzow, James A.; Solomon, Otis M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.7.1 - 45.7.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-82-K-0257; DE-AC04-76DP-00789; Copyright; Avail.: Other Sources
An algebraic procedure for linear system identification using empirically obtained noise contaminated, excitation - response data is proposed. This method entails an eigen analysis of a combined excitation - response correlation matrix estimate. This analysis provides both model order information as the required model parameter estimates. Author
System Identification; Algebra; Parameter Identification; Linear Systems; Excitation; Correlation
20060008803 Korea Advanced Inst. of Science and Technology, Seoul, Korea, Republic of
ON the CONVERGENCE BEHAVIOR of FREQUENCY-DOMAIN LMS ADAPTIVE DIGITAL FILTERS
Lee, Jae Chon; Un, Chong Kwan; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.4.1 - 46.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
In this paper we analyze the convergence behavior of the frequency-domain LMS(FLMS) adaptive digital filter(ADF) using discrete Fourier transform. We obtain the optimum Wiener solution, the minimum mean-squared-error(MSE), the convergence condition, and the excess MSE in the steady state. Also, we study the convergence behavior of the FLMSADF-based on the concept of a self-orthogonalizing algorithm. By approximating the input autocorrelation matrix as a circulant matrix, we compare the results of the LMS and FLMS ADF's. Author
Adaptive Filters; Digital Filters; Convergence; Autocorrelation; Mean Square Values; Discrete Functions
20060008804 Colorado State Univ., Fort Collins, CO, USA
PHASE UNWRAPPING VIA MEDIAN FILTERING
Loeffler, C. M.; Leonard, R. E., Jr.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.8.1 - 46.8.3; In English; See also 20060008748 Contract(s)/Grant(s): NSF ECS-83-40040; Copyright; Avail.: Other Sources
A simple phase unwrapping algorithm is presented herein. The algorithm is formulated as non-linear system through which a sampled version of the phase is passed as a signal. The system is composed of three parts, a length 2 linear FIR filter, followed by a median filter, and a 1st order IIR filter. It is also noted that this system is an extension of a median filter in that it will remove both unit impulses and unit steps. Author
Fir Filters; Linear Filters; Impulses; Algorithms
20060008828 Georgia Inst. of Tech., Atlanta, GA, USA
AN ITERATIVE METHOD FOR RESTORING NOISY BLURRED IMAGES
Katsaggelos, A. K.; Biemond, J.; Mersereau, R. M.; Schafer, R.W.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.2.1 - 37.2.4; In English; See also 20060008748 Contract(s)/Grant(s): DAA29-81-K-0024; Copyright; Avail.: Other Sources
This paper introduces a new iterative image restoration method which is capable of restoring noisy, blurred images by incorporating a priori knowledge about the image and noise statistics into the iterative procedure. the iteration equation consists of a prediction part which is based on a noncausal image model description and an innovation part which is weighted by a gain factor. The gain is computed using a linear MSE optimization procedure and is updated at each step of the iteration. This image restoration scheme can be interpreted as an iterative procedure with a statistical constraint on the image data. Author
Iteration; Restoration; Imaging Techniques; Amplification
20060008829 FMC Corp., Santa Clara, CA, USA
NONSTATIONARY 2-D RECURSIVE RESTORATION of IMAGES WITH SIGNAL-DEPENDENT NOISE
Kuan, Darwin T.; Sawchuk, Alexander A.; Strand, Timothy C.; Chavel, Pierre; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.4.1 - 37.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A nonstationary 2-D recursive image restoration filter that uses a nonstationary mean, nonstationary variance (NMNV) image model and minimizes the local mean square error is developed. The 2-D recursive filter adapts itself to the local image statistics and is able to do space-variant processing. The NMNV image model has a simple dynamic representation which simplifies the filter structure considerably. However, the optimal recursive filter still requires extensive computation. A suboptimal approach that uses a reduced update concept is proposed to reduce the computational efforts. With some modifications, this nonstationary 2-D recursive filter is extended to a class of uncorrelated, signal-dependent noise such as multiplicative noise and Poisson noise. The explicit filter structures and simulation results for images degraded by these signal-dependent noises are presented. Author
Image Filters; Restoration; Variance (Statistics); IIR Filters; Space Processing
20060008830 Utah Univ., Salt Lake City, UT, USA
SINGULAR VALUE DECOMPOSITION, SINGULAR VECTORS, and THE DISCRETE PROLATE SPHEROIDAL SEQUENCES
Zhou, Y.; Rushforth, C. K.; Frost, R. L.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 37.6.1 - 37.6.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
We study in this paper a discrete-time, discrete-frequency model for image restoration using the singular value decomposition of the imaging matrix. We show Chat the resulting singular vectors have many of the properties possessed by Slepian's discrete prolate sphroidal sequences (DPSS). They are doubly orthogonal, they are bandlimited, they satisfy an equation very similar to that satisfied by the DPSSs, and they possess an extremal energy-concentration property. These properties continue to hold with appropriate modification for bandpass as well as low-pass operations. Author
Sequencing; Decomposition; Mathematical Models; Imaging Techniques; Frequencies
20060008833 Rhode Island Univ., Kingston, RI, USA
Invariant Detection of Transient ARMA Signals with Unknown Initial Conditions
Kay, Steven M.; Scharf, Louis L.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.4.1 - 38.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A large class of physical signals may be characterized by mode parameters (natural frequencies and damping coefficients) and initial conditions. The mode parameters, usually governed by well understood physical models, are often accurately known apriori. Conversely, very little is usually known about initial conditions. The problem of interest here is to detect signals with known modes, but unknown initial conditions, in additive Gaussian noise of unknown level. The signals may also be characterized as impulse responses of ARNA systems with unknown MA parts. We derive an F-statistic that is optimum (uniformly most powerful) in the class of all receivers that are invariant to certain transformations of the data. We argue that the invariances are natural constraints. The statistic we derive provides constant false alarm rate performance. Author
Resonant Frequencies; Signal Detection; Random Noise; Noise Intensity; Receivers
20060008834 Bedford Research Associates, Inc., Bedford, MA, USA
A QUANTITATIVE COMPARISON of GENERALIZED CORRELATOR WINDOW FUNCTIONS IN PRESENCE of STRONG SPECTRAL PEAK FOR SPATIALLY SEPARATED SOURCES
Boucher, Ronald E.; Hassab, Joseph C.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.6.1 - 38.6.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Simulation results are presented to compare the effectiveness of various windowing functions for the generalized correlator when a strong spectral peak or sinusoid is present. In this case, the time delay of the desired source and of the interfering sinusoid are allowed to be different. The results show that all windows except HBII and SCOT deteriorate markedly. These two windows remain robust to the ambiguity problems presented by the sinusoid, independent of whether the spectral peak is included in the signal spectrum or in the noise spectrum. Also, window HBII performs better than SCOT in terms of probability of correct detection, variance of time delay, and SNR threshold. Author
Correlators; Noise Spectra; Sine Waves; Time Lag; Probability Theory
20060008835 Keystone Computer Associates, Inc., Fort Washington, PA, USA
THE COMPLETE CHARACTERIZATION of A MULTIPLE SINUSOID SIGNAL
Horwedel, Mark S.; Rao, Sathyanarayan S.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 38.7.1 - 38.7.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Many problems of practical import can be formulated mathematically as problems in signal identification, where the signal is modeled as o finite number of discrete sinusoidal components which are additively combined and corrupted by an additive zero mean white noise source. It is desired to identify the parameters which characterize the signal; that is, the power (or amplitude), phase, and frequency of each of the sinusoide, and the noise power. A recently developed technique which shows much promise in its ability to estimate parameters of such a signal is Pisarenko's Harmonic Decomposition (PHD). Given such a signal, PHD will estimate the number of sinusoids present, the powers of each of the sinusoids, and the power of the noise. What is missing from s complete characterization of the signal is an estimate of the phases of the sinusoids. The goal of our effort has been to extend the PHD to yield phase information and thus to give the complete characterization of the signal. Author
Noise Generators; Sine Waves; White Noise; Frequencies
20060008836 Fairchild Lab. for Artificial Intelligence Research, Palo Alto, CA, USA
SCALE-SPACE FILTERING: A New Approach To Multi-Scale Description
Witkin, Andrew P.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 39A.1.1 - 39A.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The extrema in a signal and its first few derivatives provide a useful general purpose qualitative description for many kinds of signals. A fundamental problem in computing such descriptions is scale: a derivative must be taken over some neighborhood, but there is seldom a principled basis for choosing its size. Scale-space filtering is a method that describes signals qualitatively, managing the ambiguity of scale in an organized and natural way. The signal is first expanded by convolution with gaussian masks over a continuum of sizes. This 'scale-space' image is then collapsed, using its qualitative structure, into a tree providing a concise but complete qualitative description covering all scales of observation. The description is further refined by applying a stability criterion, to identify events that persist of large changes in scale Author
Range (Extremes); Derivation; Continuums
20060008838 California Univ., Irvine, CA, USA
SPECTRAL ESTIMATION USING LEVEL-CROSSING DATA
Hosteter, Gene H.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 39B.1.1 - 39B.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Conventional signal sampling is with uniform time-step, variable amplitude. This paper considers level-crossing samples, in which the amplitude is uniform and the time-step is variable. Time-varying spectral observer techniques are applied to the problem of determining the spectrum of a suitably bandlimited signal from its level-crossings. Author
Data Sampling; Crossings; Time
20060008840 Orincon Corp., La Jolla, CA, USA
AUTOMATED DETECTION IN MULTIPLE-TARGET ENVIRONMENTS USING the CENSORED MEAN-LEVEL DETECTOR
Presley, Joe A., Jr.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 39B.3.1 - 39B.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper presents performance results for a class of robust, constant-false-alarm-rate (CFAR) detectors known as censored mean-level detectors (CMLD). CMLD have a generalized maximum likelihood detector structure that has been modified to provide robust performance in multiple target environments. In the CMLD, a censoring of selected order statistics is used in the noise power estimate to achieve the desired robustness. In the absence of interfering targets, the performance of the CMLD approaches that of the optimal parametric energy detector as M, the number of noise reference samples per test sample, approaches infinity; but even for M as small as 32, the SNR performance is typically within 0.1 dB of optimum. The two realizations of the CMLD compared in this paper are those in which either the censoring is performed before the time averaging of the noise reference samples (CBA) or the censoring is performed after the time averaging (CAA). The results indicate that censoring up to 75% of the noise reference samples significantly improves the CMLD's robustness against interference while making no significant change in the performance in the absence of interference. In addition, the results indicate that the CAA-CMLD is much more robust and computationally efficient than the CBA-CMLD. Author
Detection; Maximum Likelihood Estimates; Statistical Analysis; Time Signals; Robustness (Mathematics); False Alarms
20060008841 Naval Ocean Systems Center, San Diego, CA, USA
HIGH-RESOLUTION TECHNIQUES FOR TWO-DIMENSIONAL ESTIMATION of ANGLE-of-ARRIVAL FOR PLANAR ARRAYS
Alsup, James M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 39B.3.1 - 39B.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Two-dimensional beamforming can be a valuable technique for obtaining pure-mode signals in a multipath environment, if adequate resolution is available in both elevation and azimuth dimensions. Early simulations have shown that, for adequate signal/noise ratios and for selected signal and noise types, two sources separated by as little as .01 conventional beamwidth can be resolved. Thus, current arrays may still have adequate aperture for resolving such multipaths. The MUSIC algorithm and related eigenvalue/eigenvector analysis techniques appear to be the key to allowing the kind of superior performance which would eventually lead to an order-of-magnitude improvement in source-location accuracy. Author
Beamforming; Multipath Transmission; Signal to Noise Ratios; High Resolution; Beams (Radiation); Algorithms; Spectrum Analysis
20060008842 Yale Univ., New Haven, CT, USA
THE EFFECT of AN AUXILIARY SOURCE ON the PERFORMANCE of A RANDOMLY PERTURBED ARRAY
Ashok, Erramilli; Schultheiss, Peter M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.1.1 - 40.1.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-80-C-0092; Copyright; Avail.: Other Sources
A fundamental limitation in underwater source localization procedures using arrays of spatially distributed sensors is the uncertainty in sensor positions. This paper examines delay estimation in the presence of background noise, random array perturbations and an auxiliary source at a known location. Cramer-Rao bounds on the delay estimate are given, and the behavior of the bound as a function of the auxiliary source strength is examined. It is shown that the auxiliary source can be used to reduce estimation errors due to sensor displacements; at the same time it may add to the errors due to the background noise. It is concluded that under some conditions the auxiliary source can be u3ed to reduce the total error in the delay estimate. Author
Background Noise; Error Analysis; Perturbation; Cramer-Rao Bounds
20060008845 Virginia Polytechnic Inst. and State Univ., Blacksburg, VA, USA
A CONVERGENCE ANALYSIS of AN ADAPTIVE UNDERWATER PASSIVE TRACKING SYSTEM
Mose, Richard L.; Caputi, Maura J.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.6.1 - 4.6.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-77-C-0164; Copyright; Avail.: Other Sources
Much of the research done in the target tracking field has centered on the development of target models and associated digital filtering algorithms. One common approach has been to model target dynamics in rectangular coordinates and then use associated extended Kalman filters (EKF) to process measurement data. The rectangular coordinate system produces linear state equations, but in passive underwater tracking scenarios the measurements are typically nonlinear functions of functions of the state variables. These measurement nonlinearities necessitate an EKF. Unfortunately, EKF's often provide poor estimates or even diverge under adverse conditions such as an abrupt target maneuver, or poor target modeling mismatch. In order to eliminate this problem an adaptive tracking system has been developed that makes use of both polar coordinates and a nonlinear filtering system. This estimation technique generates a set of linear measurement equations of the polar state variables (range, depth, etc.) that we would like to estimate without the required manipulations involved in extended Kalman filtering routines. In order to test the 'robustness' of the adaptive state estimator, or tracking system, the target is free to maneuver both in velocity and depth in the first study which is of two dimensional target motion in the vertical ocean plane, and in velocity and bearing in the second which deals with the two dimensional horizontal plane. An additional benefit is that range, depth and bearing estimation is basically decoupled into separate channels of smaller dimension which allows for large computational savings over the EKF expansion techniques. Figure (I) shows the vertical plane estimator structure. Derived from text
Tracking (Position); Digital Filters; Algorithms; Polar Coordinates; Cartesian Coordinates; Kalman Filters; Equations of State; Linear Equations
20060008846 Rice Univ., Houston, TX, USA
OPTIMAL LINEAR ARRAYS FOR NARROW-BAND BEAMFORMING
DeGraff, Stuart R.; Johnson, Don H.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.8.1 - 40.8.4; In English; See also 20060008748 Contract(s)/Grant(s): N00014-K-81-K-0565; Copyright; Avail.: Other Sources
The exact equations governing the narrow-band resolution and detection capabilities of conventional (CONV) and minimum energy (ME) adaptive beamforming [1], are the basis for a class of linear arrays which offers simultaneously optimum resolution and detection capacities. Signal processing considerations indicate that, in this class, linear minimum redundancy (LMR) arrays [2] are best suited to CONV beamforming, and linear minimum hole (LMH) arrays [3] are best suited to minimum energy (ME) beamforming. The effect of coarray redundancies and holes on sidelobe level is examined analytically, and lower bounds are established on the number of coarray redundancies and holes. A simple recursive algorithm for the design of LMH arrays is presented. Author
Beamforming; Linear Arrays; Signal Processing; Narrowband; Redundancy
20060008847 Defence Research Establishment Atlantic, Dartmouth, Nova Scotia, Canada
MODAL DECOMPOSITION FOR DETECTION of PLANE WAVES WITH A LINE ARRAY RECEIVER
Walker, R. S.; Ashley, A. T.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.10.1 - 40.10.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The ability of parametric array processing methods to detect and locate plane wave arrivals is inherently limited by the stability of the estimates of the covariance properties of the field. This paper presents an analysis of the performance of one such method, Modal Decomposition, under the constraint of finite observation time. Four methods for obtaining estimates of the model parameters are demonstrated and the resultant parameter stability for simple field models is examined via simulation. Author
Plane Waves; Decomposition; Simulation; Receivers; Estimates
20060008848 Newcastle Univ., Australia
A NEW CLASS of BROADBAND TIME DOMAIN ELEMENT SPACE ANTENNA ARRAY PROCESSORS
Cantoi, Antonio; Er, Meng Hwa; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 40.11.1 - 40.11.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper describes a new approach to the design of broadband element space antenna array processor which can handle a variety of steering situations. The approach is applicable to array processor without pre-steering, with coarse pre-steering and exact pre-steering. Furthermore, the approach presented enables mismatch between signal model and actual signal scenario to be incorporated in the problem formulation. Analytical as well as simulation results on the new class of antenna array processors are presented. Author
Antenna Arrays; Broadband; Antennas; Simulation
20060008849 James Cook Univ. of North Queensland, Townsville, Australia
A BIT SERIAL LINEAR ARRAY DFT
Allen, Gregory H.; Denyer, Peter B.; Renshaw, David; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41A.1.1 - 41A.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A linear array which computes the DFT in a pipelined fashion is described. The algorithm is derived from the batch processing array proposed by H.T. Kung [1] but has been modified to allow continuous operation. This computation of the DFT is a complex polynomial evaluation on the unit circle using Homer's method having the data for the polynomial coefficients. Data and the N-th complex roots of unity are input at one end of the array and the DFT sequence is output from the other. The polynomial coefficients are stored in successive modules in the array and a new batch is latched successfully with a synchronizing signal. In its simplest form the design has a single system part which is replicated N times for an N-point transform. For time mutiplexed modules the system throughput and hardware can be optimized for given applications. A bit serial layout for 6 micron NMOS VLSI has been designed and simulated using the FIRST silicon compiler at the University of Edinburgh. Author
Linear Arrays; Batch Processing; Modules; Metal Oxide Semiconductors; Compilers; Algorithms
20060008850 Calgary Univ., Alberta, Canada
A BIT SERIAL LDI RECURSIVE DIGITAL FILTER
Turner, L. E.; Denyer, P. B.; Renshaw, D.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41A.3.1 - 41A3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The practical implementation of a bit serial lossless discrete integrator (LDI)[I] recursive ladder filter suitable for implementation as a single integrated circuit is described. The low coefficient sensitivity and simplicity of the LDI signal flowgraph make the filter structure suitable for implementing high quality digital filters. A bit serial integrated circuit filter has been designed and simulated using the silicon compiler system FIRST [2,3]. The LDI filter coefficients which yield a Chebyscheff transfer function characteristic are found using an exact synthesis method[4]. The finite precision time domain response of the filter is simulated using the FIRST simulator and the magnitude response is verified by calculating the fourier transform of the filter unit sample response. The LD[ filter implementation makes use of an alternate clocking scheme which simplifies the signal flow graph. This is a form of multiplexing which is easily implemented using bit serial arithmetic. Author
Digital Filters; Integrated Circuits; Signal Flow Graphs; IIR Filters; Integrators; Compilers
20060008852 Bonn Univ., Germany
MEASURING the DEGREE of NON-STATIONARITY BY USING the WIGNER-VILLE SPECTRUM
Martin, Wolfgang; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.3.1 - 41B.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The Wigner-Ville spectrum is the only conjoint time-freq,ency representation of the second order properties of non-stationary random signals which is real-valued and compatible with linear filtering and modulation. Pseudo Wigner estimators are known to form a class of estimators being more flexible than short time periodograms: They give uncorrelated estimates for appropriately spaced neighboured frequencies allowing separately to measure the degree of non-stationarity for each band of frequencies. Using the model of quasi stationary processes, their stability and precision are defined as a measure of their character of non-stationarity. The performances are then tested on simulated data and successfully applied to the problem of classifying biological time-keeping facilities. Author
Random Signals; Time Measurement; Linear Filters; Stability; Wigner Coefficient; Estimates
20060008853 Ecole Nationale Superieure des Telecommunications, Paris, France
TIME-FREQUENCY ANALYSIS USING TIME-DEPENDENT ARMA MODELS
Grenier, Y.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.5.1 - 41B.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper discusses the estimation of time-frequency representations of non-stationary signals, by means of ARMA (autoregressive moving-average) models with time-dependent coefficients. In order to allow the estimation of the model on a single realization of the random signal, it is assumed that the coefficients are weighted sums of known functions, for instance polynomials, cosines... Two ARMA models will be discussed in this paper. The first one is a white noise driven ARMAmodel, which is rather usual. The second one has been designed for deterministic or speech-like signals and consists of a linear system driven by intermittent inputs with additive output (white) noise. The time-frequency representation of the signal is then obtained from the state-space realization of the time-dependent model. After a description of the algorithms involved in both estimators, the paper focuses on e comparison of the two ARMA models on modulated sinusoids corrupted by white noise. This shows that these models behave equally well for high SNR (signal-to-noise ratio), but the deterministic model gives better results when the SNR decreases. Author
Time Dependence; Signal to Noise Ratios; Linear Systems; Autoregressive Moving Average; Frequencies; Time Signals; Random Signals
20060008854 Laboratoire Traitement du Signal et Instrumentation, Saint-Etienne, France
JOINT REPRESENTATION (JR) IN SIGNALTHEORY (ST) and HILBERTIANANALYSIS :APOWERFULTOOL FOR SIGNAL ANALYSIS
Escudie, B.; Crea, J.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.6.1 - 41B.6.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Joint Representations were derived to display signals as an energetic function of time t and frequency nu . These variables are conjugate ones as position and momentum. in Quantum Mechanics (QM). Properties of various JRs are depicted by a measurement system characterized by a weighting function f(n,t). This function is related to the moments of JR related to modulation properties of signals. Operators defined by hilbertian analysis are associated by various JRs to variables t and nu and functions of them, as in Q.M. Properties of JR are discussed as described by f(n,t). Author
Signal Analysis; Quantum Mechanics; Time Dependence; Weighting Functions; Frequencies; Modulation
20060008855 Philips Research Labs., Eindhoven, Netherlands
ON the TIME-FREQUENCY DISCRIMINATION of ENERGY DISTRIBUTIONS: CAN THEY LOOK SHARPER THAN HEISENBERG?
Claasen, T. A. C. M.; Mecklenbrauker, W. F. G.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.7.1 - 41B.7.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This contribution addresses the implications of Heisenberg's uncertainty relation on time-frequency energy distribution functions depending bilinearly on the signal. It is indicated that all these distribution functions must have a certain minimum spread in the time-frequency plane. Furthermore such distribution functions can either have correct marginal distributions or be positive, but not both. Therefore a local interpretation of these energy distributions must be taken with care. Some facts determining the choice of an appropriate distribution function will be discussed. A comparison will be made between the spectrogram and the Wigner distribution. Author
Temporal Distribution; Time Discrimination; Frequency Distribution; Distribution Functions; Spectrograms; Quantum Mechanics
20060008856 Orincon Corp., La Jolla, CA, USA
SPECTROGRAMS and GENERALIZED SPECTROGRAMS FOR CLASSIFICATION of RANDOM PROCESSES
Altes, Richard A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 41B.8.1 - 41B.8.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Amaximum likelihood (ML) classifier for discriminating between nonstationary Gaussian time series can be implemented by correlating the data spectrogram with templates that are constructed from ensemble average reference spectrograms. The time window used to synthesize the spectrograms must have a duration that is longer than the decorrelation time of the data in the neighborhood of the window. If the data time series exhibits significant nonstationarity within this decorrelation time, Karhunen-Loeve (K-L) basis functions should ideally be used to construct a generalized spectrogram, rather than using a standard spectrogram constructed with the usual sinusoidal basis functions. Utilization of a standard spectrogram imposes forced, pseudo-stationarity by approximating the autocovariance function of the data by the short-time autocorrelation function. This forced stationarity is routinely used to obtain linear prediction coefficients (LPC). When signal to interference ratio (SNR) is large, the templates that are used to classify a data spectrogram are sensitive to differences in the locations of nulls or zeroes in the expected signal spectrograms from different data classes. This null sensitivity seems to imply that peak-oriented models of random processes, e.g., the all pole representation that is associated with LPC, are suboptimum for ML classification under high SNR conditions. Compensation for time warping is especially necessary if window durations are data dependent. Spectrogram implementation of the ML classifier yields a new similarity index for time warp compensation. Author
Random Processes; Spectrograms; Autocorrelation; Karhunen-Loeve Expansion; Linear Prediction; Time Series Analysis; Time Functions; Image Classification
20060008860 Purdue Univ., West Lafayette, IN, USA
MODELING of ENGLISH SPEECH FOR the DESIGN of A DISTRIBUTED SPEECH UNDERSTANDING SYSTEM
Bronson, Edward C.; Coyle, Edward J.; Slegel, Leah Jamieson; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 42.6.1 - 42.6.4; In English; See also 20060008748 Contract(s)/Grant(s): NSF ECS-81-20896; Copyright; Avail.: Other Sources
This paper describes the derivation and verification of a phoneme model of English speech. The model is used to generate a stream of phonemically labeled speech frames to model speech input for the design of a distributed speech understanding system. New computer architectures to perform speech understanding in real time should incorporate information about the characteristics of English speech. In order to predict the performance of a new architecture, it is necessary to simulate the design using either massive amounts of speech data or, as an alternative, a statistical model of speech. A statistically generated phoneme stream is used to avoid the difficulty of performing computationally intensive acoustic parameterization on the enormous amount of speech input data which would be required to obtain representative phoneme distributions and patterns of speech. Author
Parameterization; Architecture (Computers); Speech Recognition; Performance Prediction; Phonemes; Mathematical Models
20060008862 Nancy Univ., France
AN EXPERT SYSTEM FOR the AUTOMATIC READING of FRENCH SPECTROGRAMS
Carbonella, N.; Fohr, D.; Haton, J. P.; Lonchamp, F.; Pierrel, J. M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 42.8.1 - 42.8.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
We report on a long term project on the phonetic decoding of continuous speech, within the framework of an expert system. Spectrogram reading is the field of expertise under study, although it is clear that an automatic phonetic decoder should not be in the form of an expert system. We expect major improvements in the recognition rate when expert's rules will be incorporated into existing systems. We describe here both the method used to retrieve the human expertise and results by an experienced spectrogram reader on 50 phonetically-balanced sentences.We also sketch various ways in which the expert rules can be validated through an 'inference engine' and actually used in a system. Finally, we discuss the impact this approach has on the design. Author
Expert Systems; Phonetics; Spectrograms; Decoders; Readers
20060008864 National Technical Univ., Athens, Greece
EFFICIENT ALGORITHMS and STRUCTURES FOR LAGGED LEAST SQUARES(LS) FIR FILTERS IN the CASE of PREWINDOWED SIGNALS
Carayannis, G.; Manolakis, D.; Kalouptsidis, N.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 43.5.1 - 43.5.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The purpose of this paper is to provide a brief overview of the algorithms and structures which can be used in connection with l- lag FIR filtering. A number of existing efficient algorithms from the zero - lag problem are generalized here, to be applied to the l-lag case. Both direct and lattice - ladder structures are considered. Four different families of techniques are discussed: time recursive, order recursive, lattice - ladder and ' recursive in lag'. The unified approach used in this paper for the derivation of the algorithms permits the underlining of the relationships among the variables appearing in the four families of recursive schemes discussed. The 'prewindowing' assumption is used throughout. Author
Fir Filters; Errors; Algorithms; Predictions; Time Signals
20060008867 Texas Instruments, Inc., Houston, TX, USA
A STEP TOWARD REAL-TIME INTERACTIVE FIR FILTER DESIGN
Simar, Ray, Jr.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.1.1 - 44.1.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper outlines the development of a new tool for the digital signal processing engineer. The coupling of a frequency sampling technique for the design of linear-phase FIR filters with recent advances in digital signal processors has led to the development of a real-time interactive FIR filter design technique. The system accepts samples of the desired frequency response as inputs, determines the corresponding linear-phase FIR filter, and implements the filter. This allows the DSP engineer to adjust the filter while simultaneously examining its response to an input signal. The input signal might be a test signal derived from a spectrum analyzer, an audio signal, or any general signal source of interest. This flexibility provides the user with what will hopefully be a valuable tool in a broad variety of digital signal processing applications. The simplicity and utility of the system is further emphasized by the implementation of the system on a single Texas Instruments TMS32010 digital signal processor. Author
Digital Systems; Fir Filters; Real Time Operation; Signal Processing; Electrical Engineering
20060008868 TRW LSI Products, La Jolla, CA, USA
NEW CMOS CHIP FACILITATES MULTIBIT CORRELATION
Eldon, J. A.; Haight, J. D.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.2.1 - 44.2.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Although correlation is a simple mathematical function, its implementation in digital hardware is cumbersome and complex. During the past few years, both digital and analog correlator integrated circuits have helped reduce the cost and size of correlation hardware. Unfortunately, the analog correlators exhibit noise and calibration problems, while the digital devices are generally limited to single-bit resolution, requiring the parallel use of many devices or the use of very coarse signal quantization levels. In response to these objections and to the need for low power correlators, TRWLSI Products is developing the TMC2220, a four-channel, 32-word CMOS digital correlator chip. This paper outlines the architecture of this versatile chip and illustrates some of its advantages in multibit and dual-channel (in phase and quadrature) applications. Potential noise performance of the fourbit correlator is compared to that of similar single-bit and ideal analog devices. A few applications which might best exploit the TMC2220's architecture are also discussed. Author
CMOS; Correlators; Chips (Electronics); Digital Systems
20060008869 Speech Plus, Inc., USA
A Low Cost FFT Chip Set
Sweitzer, Steve; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.3.1 - 44.3.3; In English; See also 20060008748; Copyright; Avail.: Other Sources
The current generation of LSI signal processors has opened a whole range of new commercial applications using real time signal processing. Among the more attractive of these areas are speech compression, speech recognition, telephone line conditioning, and voice grade modems. This paper illustrates the feasibility of implementing an FFT on voiceband signals in real time. A compact, low cost chip set comprised of a Texas Instruments TSM320 signal processor, two 2kx8 CMOS RAMS, plus a couple of standard MSI components has been designed to perform a 1024 point complex FFT. Intitial analysis indicates that the system executes the transform in 78msec, a rate commensurate with real time processing at sampling rates up to 12.8kHz. The total cost of the components, in volume, is approximately $150, and is likely to decline as the TMS320 fabrication process traverses the learning curve. Author
Chips (Electronics); Fast Fourier Transformations; Large Scale Integration; Low Cost; Signal Processing; Fabrication
20060008870 Nippon Electric Co. Ltd., Kawasaki, Japan
A CMOS-VLSI RATE CONVERSION DIGITAL FILTER FOR DIGITAL AUDIO SIGNAL PROCESSING
Hirosaki, B.; Tomimitsu, Y.; Ishihara, S.; Nakada, H.; Akiyama, K.; Nosaka, K.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.4.1 - 44.4.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper describes CMOS-VLSI architecture for a rate conversion digital filter which interpolates an input discrete-time audio signal at the twice times higher sampling rate. In the proposed architecture, two discrete-time audio signals with 16-bit word length are multiplexed into a serial bit stream and are simultaneously interpolated through parallel table look up array multiplications. Based on the architecture study, the VLSI interpolator which contains 34,000 transistors has been fabricated with 3 micrometer CMOS technologies. Its power consumption is less than 100 mW and the maximum input sampling rate is 50 kHz. The interpolator quality is specified in terms of inband amplitude ripple and out-band attenuation each of which is 0.13 dB or 80 dB, respectively. Author
Audio Signals; CMOS; Digital Filters; Fabrication; Signal Processing; Very Large Scale Integration
20060008872 Stemens A.G., Munich, Germany
MOS-VLSI PIPELINED DIGITAL FILTERS FOR VIDEO APPLICATIONS
Ulbrich, W.; Noll, T.; Zehner, B.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.7.1 - 44.7.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The interactions between digital filter design and VLSI design methodology are discussed. The constraints imposed by the MOS-architecture are discussed, an appropriate filter structure is proposed. The mapping of this structure to a VLSI realization is shown for intermediate and high sample rates. Design examples for FIR and IIR filtes for video applications and lowpass filters for decimation and interpolation are given. Author
Digital Filters; Very Large Scale Integration; Video Communication; Electrical Engineering; Metal Oxide Semiconductors
20060008873 Thomson-CSF, Orsay, France
CIRCUITS FOR DIGITAL SIGNAL PROCESSING
Barral, H.; Moreau, N.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.9.1 - 44.9.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
This paper discusses two custom integrated circuits designed to perform the functions of signal correlation and lattice filtering (HA or AR). Each circuit is decomposed into P operators, each being a direct implementation of the equations. To allow concurrent use of an arbitrary number of operators and to simplify inter-module connections (both within and between chips), a bit-serial architecture was adopted. These chips can be cascaded; computation speed is independent of model order in both types of calculations. These chips have been designed to operate at a sample frequency between 0 and 300 kHz for the correlator, 0 and 150 kHz for the lattice filter. Author
Circuits; Signal Processing; Computation; Application Specific Integrated Circuits; Chips
20060008874 Florida Univ., Gainesville, FL, USA
A LOGARITHMIC ARITHMETIC UNIT FOR SIGNAL PROCESSING
Taylor, Fred J.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 44.10.1 - 44.10.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The logarithmic number system, or LNS, has the ability to support very high-speed arithmetic, over a wide dynamic range, in limited hardware. This system, with extensions, has several architectural and data enhancement attributes that makes it attractive to the DSP scientists and engineer. In this paper, the regularity of the LNS, its data compression capabilities, and speed complexity potential are advocated in a DSP setting. It Is shown that the LNS, and a floating point variation on this theme, offer distinct advantages over classical DSP architectures. Author
Arithmetic and Logic Units; Signal Processing; Logarithms
20060008878 Technion - Israel Inst. of Tech., Haifa, Israel
EFFICIENT NONLINEAR SYSTEM IDENTIFICATION
Mansour, David; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.3.1 - 45.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
System identification of a second order truncated Volterra series with correlated and Gaussian input is investigated. This problem has been treated by Schetzen using Wiener nonlinear theory. In this paper we show how this nonlinear system can be efficiently identified using Gaussian properties of the input. In a second order Volterra representation there are N unknowns elements for the linear kernel and an additional N(sup 2) unknowns elements representing the second order Volterra kernel. The identification of the system by standard least squares technique require to solve a set of 1/2N(N+3) linear equations, or equivalently to invert a matrix of dimension 1/2N(N+3). Using the fact that for Gaussian signals all the higher moments are determined by the first two, we show that the identification of the second order truncated Volterra series can be reduced to an inversion of a matrix of dimension N+1. An additional advantage of the proposed method is that is is applicable to any correlated Gaussian signals; Schetzen method is limited to correlated Gaussian process generated by invertible filters. Author
System Identification; Linear Equations; Nonlinear Systems; Nonlinearity
20060008880 Edinburgh Univ., UK
NON-LINEAR SYSTEM MODELLING: CONCEPT and APPLICATION
Cowan, C. F. N.; Adams, P. F.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.6.1 - 45.6.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The existing theory relating to the analysis and modelling of non-linear systems relies on the Wiener mode] which is unnecessarily complex in many practical situations. This paper presents a non-linear system modelling technique based on the 3-section block mode] which may be reconfigured to represent the non-linearities present in many practical situations. Author
Nonlinearity; Architecture (Computers); Digital Systems; Signal Processing; Digital to Analog Converters; Adaptive Filters
20060008881 Princeton Univ., NJ, USA
An Iterative Algorithm for Locating the Minimal Eigenvector of a Symmetric Matrix
Fuhrmann, Daniel R.; Liu, Bede; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 45.8.1 - 45.8.4; In English; See also 20060008748 Contract(s)/Grant(s): AFOSR-81-0186; Copyright; Avail.: Other Sources
A new iterative method of finding the minimum eigenvalue of a symmetric matrix is described. This method does not utilize matrix inversions and is applicable to any matrix R for which the matrix-vector product Rx is rapidly computable. It seeks the minimum eigenvalue of R by minimizing the quadratic form x(sup T)Rx on the unit hypersphere, using a search technique derived from the conjugate gradient method. The computational complexity of each step of the algorithm depends on the speed with which Rx can be computed. Author
Iteration; Algorithms; Matrices (Mathematics); Matrix Methods; Inversions; Eigenvalues
20060008882 Sanders Associates, Inc., Nashua, NH, USA
CORRELATION of INTERPOLATED TIME DELAYED COMMUNICATION SIGNALS
Struckman, Keith A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.3.1 - 46.3.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Correlation of a narrow bandwidth communication signal, uniformly sampled at separate receive sites, yields time-difference-of-arrival (TDOA) resolutions to one-half the sample-to-sample spacing. An interpolation scheme must be employed if smaller resolutions are required. Interpolation may be applied to the correlation envelope or as analyzed in this paper to intrasample shift of one of the complex time waveforms. The magnitude of the correlation and position of the shift at the correlation maximum provide a quantitative measure of the interpolation scheme. The magnitude of the correlation coefficient is converted to an equivalent signal-to-noise (ESNR) for comparisons with conventional SNR measurements. The data presented uses interpolated samples that were generated by a short sum of adjacent sample sine function products. Author
Correlation Coefficients; Signal to Noise Ratios; Time Response; Time Signals; Interpolation; Waveforms
20060008885 British Columbia Univ., Vancouver, British Columbia, Canada
COMMENTS ON PARAMETRIC and NON-PARAMETRIC DETECTION of EPILEPTIFORM SPIKE ACTIVITY
Beddoes, M. P.; Wada, J. A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.7.1 - 46.7.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
Parametric methods which will detect spikes in EEG signal have been suggested by many authors, and encouraging results seem to have been obtained. A main limitation is instrument complexity and most of the results have been obtained relatively slowly off line. The basic theoretical limitations of the approach apart from the factor of instrumentation do not seem to be fully recognized. In this paper we show 1)that the parametric method is not optimum in where the Weiner sense. 2) that anamolous indications of spike activity are given when p, the number of filter parameters, is varied. 3) that the mean square error is not an indicator of performance. 4) that low p values, perhaps surprisingly, lead to better spike detection than high values, and Non-parametric methods have been suggested by less instrumentation is required and on-line results can be obtained in real time using modest equipment.A16-channel real-time spike monitor has been realized on an Apple II computer using a method attributed to Ninomija et al. The basic approach is to detect incidence of shapes, all of which could be actual spikes, and then in subsequent filtering stages remove those shapes which do not meet the criteria spelled out by the neurosurgeon. Author
Electroencephalography; Mean Square Values; On-Line Systems; Real Time Operation; Errors
20060008886 Princeton Univ., NJ, USA
Adaptive Equalizer Using Finite-Bit Power-of-Two Quantizer
Xue, Ping; Liu, Bede; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.9.1 - 46.9.4; In English; See also 20060008748 Contract(s)/Grant(s): AFOSR-81--186; Copyright; Avail.: Other Sources
The performance of a simplified adaptive algorithm using finite-bit power-of-two quantizer is analyzed by assuming Gaussian distributions for both input signals and equalizer output errors. The results show that in spite of its simple implementation, the performance is comparable with the least mean square (LMS) algorithm. The convergence and the stability is then studied, the output mean square error is analyzed and the effects of non-Gaussian cases are considered. Computer simulation results support the theoretical results. Author
Algorithms; Equalizers (Circuits); Adaptive Control; Counting Circuits
20060008887 General Electric Co., Syracuse, NY, USA
ADAPTIVE ARRAY CANCELLATION of MULTIPATH INTERFERENCE
Morgan, Dennis R.; Aridgides, Athanasios; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 46.10.1 - 46.10.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
The steady-state performance of a narrowband adaptive array processor using a single auxiliary tapped delay line is analyzed for a simple two-path interference scenario, and computer solutions are presented for various combinations of signal and multipath parameters. It is shown that auxiliary delay taps are required for effective cancellation of multipath interference, even for fractional bandwidth- delay products. Good cancellation is generally achieved provided that (a) the delay line taps are spaced somewhat closer than the Nyquist sampling rate, and (b) the total length of the delay line considerably exceeds the longest multipath delay, depending on multipath component strengths. A simple impulse response model is also developed to provide an understanding of the behavior of cancellation performance and how it relates to the various parameters involved. Author
Multipath Transmission; Antenna Arrays; Narrowband; Delay Lines
20060008893 Raytheon Co., Portsmouth, RI, USA
MODERN, ACTIVE SONAR AGC DESIGN CONSIDERATIONS
Seegal, Robert H.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 47.10.1 - 47.10.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
There have been tremendous strides in digital signal processing that have had impact on sonar design, ranging from the rediscovery and wide dissemination of the Fast Fourier Transform algorithm to homomorphic deconvolution techniques. Over the last two decades, compact, low power digital circuits have brought reliable, multichannel signal processing capabilities to sonar. Unfortunately, the signals that we want to analyze are analog. Today, analog-to-digital converter systems are available in a package the size of a matchbook. Even this is not enough. The signals that an active sonar has to deal with often exceed the dynamic range of even a 14-bit A/D converter. Consequently, the A/D converter in a sonar system is preceded by a variable gain amplifier. A generic sonar system is shown. The step-gain amplifier and its controller are the variable gain element. The beamformer includes an A/D converter. The gain has to be adjusted in such a way that the incoming signal fits within the dynamic range of the A/D converter. If the signal level is too high, the signal will be badly distorted by clipping; if the signal level is allowed to drop too low, quantization noise degrades processing. The gain control element has to adjust the variable gain amplifier control so that clipping distortion and quantization noise are minimized. In an active sonar, this has to be done even though the signal level is varying rapidly. Because sonars have typically on the order of a hundred signal channels, as many amplifiers are required, for these analog circuits cannot be multiplexed. To meet channel-to-channel gain tracking requirements, step-gain amplifiers serve as the gain adjusting element. This paper is concerned with how the gain should be controlled for an echo-ranging sonar. Derived from text
Sonar; Automatic Control; Amplification; Electrical Engineering
20060008900 Grenoble-1 Univ., Saint Martin d'Heres, France
OPTIMAL DATA ESTIMATION SYSTEM OVER MULTIPATH CHANNEL
Jourdian, G.; Martin, J.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 24.11.1 - 24.11-4; In English; See also 20060008748; Copyright; Avail.: Other Sources
We are interested in optimal linear estimation of signals transmitted over a noisy multipath channel. The impulse response of the channel is assumed known. The problem is similar to the telecommunication equalization problem. The optimal linear discrete causal and stable filter (aWiener-type filter) is given; different structures are studied and their performance calculated: i) a pure transversal structure, ii) a mixed recursive and transversal structure which is simple and well-adapted to the model of channel. Some results are exhibited. This last structure has been found robust enough versus an error on the channel estimated parameters. Author
Multipath Transmission; Channels (Data Transmission); Noise (Sound); Mathematical Models
20060008901 Belgrade Univ., Yugoslavia
ESTIMATION of FREQUENCIES of SINUSOIDS USING the EXTENDED KALMAN FILTER
Stankovic, Srdjan S.; Dragosevic, Marina; Carapic, Miodrag; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 5.11.1 - 5.11-4; In English; See also 20060008748; Copyright; Avail.: Other Sources
In this paper the application of the extended Kalman filter to the estimation of frequencies of sinusoids in the additive colored noise is proposed. Starting from convenient state-space models of the signal two versions of the algorithm are defined. Comparisons with both the generalized least-squares and the maximum likelihood method show its advantages provided adequate initial conditions are ensured. Author
Kalman Filters; Sine Waves; Frequencies; Maximum Likelihood Estimates; Algorithms
20060008902 Tokyo Univ., Japan
AUTOMATIC RECOGNITION of SPOKEN WORDS FROM A LARGE VOCABULARY USING SYLLABLE TEMPLATES
Fujisaki, Hiroya; Hirose, Keikichi; Inoue, Tomohiro; Sato, Yasuo; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '84); Volume 3; 1984, pp. 26.12.1 - 26.12.4; In English; See also 20060008748; Copyright; Avail.: Other Sources
A system is proposed for automatic speech recognition using syllable templates. In this system, input speech signal is analyzed and matched against syllable templates and converted into parameters characterizing each candidate syllable. Word recognition is based on an overall likelihood measure calculated for each item stored in the lexicon. A method is also developed for the optimization of syllable templates. The validity of the proposed method was tested in a preliminary recognition experiment in which a lexicon consisting of 1000 city names was used to recognize utterances of 100 city names by a female speaker. The rate of correct recognition was 96.5%. Author
Speech Recognition; Speech; Words (Language); Templates; Syllables
Source: NASA
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