Showing papers on "Adaptive beamformer published in 1981"
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TL;DR: In this paper, the ability of a least mean square (LMS) adaptive array to adapt to the electromagnetic polarization of incoming signals is considered and the output signal-to-interference-plus-noise ratio (SINR) from the array is computed as a function of the signal angles of arrival and polarizations.
Abstract: The ability of a least mean square (LMS) adaptive array to adapt to the electromagnetic polarization of incoming signals is considered. An array of two pairs of crossed dipoles is studied. A desired signal and an interference signal are assumed to arrive from arbitrary directions with arbitrary elliptical polarizations. The output signal-to-interference-plus-noise ratio (SINR) from the array is computed as a function of the signal angles of arrival and polarizations. It is shown that as long as certain special desired signal polarizations are avoided, the array is difficult to jam with a single interference signal. To produce a poor SINR, an interference signal must both arrive from the same direction and have the same polarization as the desired signal.
109 citations
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TL;DR: In this paper, the limitations of adaptive beamforming are discussed in terms of lack of a priori knowledge of source distribution and medium propagation parameters, and the underlying assumptions of classical directivity pattern application are identified.
Abstract: Underlying assumptions of classical directivity pattern application are identified. Adaptive beamforming principles are related to relaxation of these assumptions, and the limitations of adaptive beamformers are developed in terms of lack of a priori knowledge of source distribution and medium propagation parameters. Progress in spatial processing beyond current adaptive beamforming is discussed in terms of estimating jointly source distribution and medium properties using the outputs of an N‐element array of sensors.
17 citations
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TL;DR: In this article, a four element adaptive array using a modulated pilot signal added to the communication signal at the transmitter has been constructed and tested at high frequency (HF) in order to discriminate against undesired HF multipath.
Abstract: A four element adaptive array using a modulated pilot signal added to the communication signal at the transmitter has been constructed and tested at high frequency (HF). The pilot signal modulation was designed to discriminate against undesired HF multipath. The array reduces the strength of undesired modes by spatial nulling, while maintaining response in the direction of the desired mode. The array was used to receive signals that propagated via a 150-mi over-ocean path. The antenna configuration and frequency selection were such that the ground wave and ionospheric modes of propagation were approximately equal. The pilot signal, used as a reference for the adaptive array, consisted of a single tone, phased reversed by a pseudorandom sequence. The bandwidth of the pilot signal was selectable at either 3 kHz or 6 kHz. A pulse sounder was used to measure the response of the array system to the arriving modes. During the test the ground wave 1E, 1F_{1} and 1F_{2} modes were observed, occasionally simultaneously. The array reference signal could be locked to any arriving mode and the array processor was able to discriminate against all other modes by directing spatial nulls. The reduction in strength of the undesired modes measured during the test was greater than 15 dB.
12 citations
01 Aug 1981
TL;DR: In this paper, an adaptive beamforming method is proposed to improve the bearing resolution of a passive array, having many similarities to the minimum energy approach, where the evaluation of energy in each steered beam is preceded by an eigenvalue-eigenvector analysis of the empirical correlation matrix.
Abstract: A method of improving the bearing-resolving capabilities of a passive array is discussed. This method is an adaptive beamforming method, having many similarities to the minimum energy approach. The evaluation of energy in each steered beam is preceded by an eigenvalue-eigenvector analysis of the empirical correlation matrix. Modification of the computations according to the eigenvalue structure results in improved resolution of the bearing of acoustic sources. The increase in resolution is related to the time-bandwidth product of the computation of the correlation matrix. However, this increased resolution is obtained at the expense of array gain.
10 citations
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TL;DR: In this paper, a beamforming system was proposed for a horizontal line array with an element-sited beamformer, which is coupled to an array cable of a HAP.
Abstract: This invention is a hydrophone (element) sited beamformer which is coupled to an array cable of a horizontal line array. A hydrophone-sited beamformer is coupled to each hydrophone in the array. The element-sited beamformer directly forms beams from the data detected by the hydrophones by delaying and selecting some of the hydrophone detected data and summing this data with the proper shading value.
6 citations
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TL;DR: The Three-Array Processor (TAP III) beamforming system incorporating both wide-band time- domain beamforming and narrow-band frequency-domain beamforming is described, and it is shown how the fast Fourier transform (FFT) is applied to accomplish frequency- domains beamforming.
Abstract: The Three-Array Processor (TAP III) beamforming system incorporating both wide-band time-domain beamforming and narrow-band frequency-domain beamforming is described. This paper briefly develops the beamforming theory and shows how the fast Fourier transform (FFT) is applied to accomplish frequency-domain beamforming. The frequency-domain beamformer operates in the frequency domain to form beams and power spectrum data over narrow frequency bands of interest. A real-time digital filtering technique is used to extract the narrow bands of interest from the broad-band input signal. The frequency-domain beamformer accomplishes real-time digital filtering and beamforming by using a high-speed array processor to do the complex calculations and data handling required by the algorithm. The time-domain beamformer operates in parallel with the frequency-domain beamformer to form up to 16 broad-band beams in the time domain. A programmable all-pass digital filter is used to create the fine time delays required by the time-domain beamformer.
6 citations
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TL;DR: It is proved that the adaptive array acts to reduce the undesired noise power at the array output, however, improvement in output signal-to-noise ratio is insufficient because of the limitation in null formation.
Abstract: In antenna arrays, the number of nulls that can be formed independently is equal to or less than the number of array elements minus one. Therefore, if the number of interferences exceeds the number of freedom for pattern synthesis in adaptive arrays, nulls in the radiation pattern can not be steered toward each interference direction. To clarify the behavior of the adaptive array in such an environment, an analysis is made for the case where two interferences are impinging on the two-element directionally constrained adaptive array. As a result, it is proved that the adaptive array acts to reduce the undesired noise power at the array output. However, improvement in output signal-to-noise ratio is insufficient because of the limitation in null formation.
5 citations
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TL;DR: Data illustrates the suppression of side lobe interference levels achieved with an adaptive beamformer DICANNE (Digital Interference Canceling Adaptive Null Network Equipment), a 32‐channel system operating broadband over a 2‐octave audio bandwidth.
Abstract: Experimental data illustrates the suppression of side lobe interference levels achieved with an adaptive beamformer DICANNE (Digital Interference Canceling Adaptive Null Network Equipment). The experimental DICANNE is a 32‐channel system operating broadband over a 2‐octave audio bandwidth. Although the experimental nulls formed were not perfect, a 5‐dB improvement in detection threshold was observed at an element interference‐to‐noise ratio of 0 dB.
5 citations
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01 Jan 1981TL;DR: The paper presents recursive algorithms for time domain array beamforming that are computationally efficient and rapidly converging and used for adaptive beamforming both for a Frost type processor with a constraints and a partitioned processor without constraints.
Abstract: The paper presents recursive algorithms for time domain array beamforming. The algorithms are computationally efficient and rapidly converging. The algorithms can be used for adaptive beamforming both for a Frost type processor with a constraints and a partitioned processor without constraints.
4 citations
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01 Dec 1981TL;DR: The application of the array approach to the class of fimte (shift) rank processes is dealt with, which can be applied to a large variety of signal processing problems.
Abstract: We present a Hilbert space array approach for deriving fast estimation and adaptive signal processing algorithms. These are obtained using, as a basic tool, projection operators and orthonormalizations, via the Gram-Schmidt procedure. The resultant algorithms are recursive in time and order, and are realized in ladder forms. The importance of this array method is that it can be applied to a large variety of signal processing problems. This paper deals with the application of our array approach to the class of fimte (shift) rank processes.
3 citations
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TL;DR: The book is well organized, discussions are clear, important points are carefully highlighted, consistent notation has been used throughout, great care has been taken with examples, and there are good collections of problems for each chapter.
Abstract: The title of the book is accurate. It is concerned much more with digital control than digital signal processing, and to this extent, has but limited overlap with the central concerns of the IEEE TRANSACTIONS ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING. To be sure, there is some elementary material on z-transforms, on digital filter equivalences to continuous-time filters, and a brief discussion of quantization and sample rate selection, but the rest of the book is fairly and squarely control systems material, and even that material which is not specifically of a control nature is flavored with overtones and examples drawn from control engineering. This is, of course, not meant as a criticism: the authors have in no way misrepresented the subject matter of the book in choosing a title. The book’s preface opens with an assertion that it is “about the use of digital computers in the real-time control of dynamic systems such as servomechanisms, chemical processes, and vehicles which move over water, land, air, or space.” What this means is that the book has as its central concern the analysis and design of sampled-data control systems, from which it branches out to discuss various specialized. themes. Illustrations are .taken from the applications areas mentioned. Other than in its mention of quantization and sampling rate problems, there is no significant discussion of the detailed computer-related problems of, for example, interfacing, choice of programming language, and the like. The preface also asserts that the book is in effect pitched at students who have had something like a first course in control systems, with some knowledge of matrix algebra but no previous acquaintance with state-space ideas. In the reviewer’s experience as an instructor of a course using the book as text, this is accurate, save that one could add that prior exposure of students to state variable ideas would be of great help. Apart from the question of background knowledge, the general level of maturity which the book calls for from students is about that which a senior should have. The book is well organized, discussions are clear, important points are carefully highlighted, consistent notation has been used throughout, great care has been taken with examples, and there are good collections of problems for each chapter. The authors have also resisted the temptation to include the more baroque details of many aspects of the subject, and have gone to some care to locate helpful reference material (various transforms, matrix theory and notions of probability and stochastic processes) in an Appendix. All this means that students find the book very attractive to use. Because it is also up to date, and because of the subject matter covered (described below), this reviewer as an instructor using the book also found it very attractive, and certainly outranking other competitors. After an introductory chapter, the book considers the z-transform description of sampled-data systems, and several approaches to finding discrete equivalents to continuous transfer functions, the latter being most carefully done. Next, the book describes the analysis of sampled data systems, followed by the design of controllers for such systems. Classical control approaches are used to this point; the discussion is comprehensive, and makes good contact with continuous time ideas which students should have experienced earlier. One could perhaps quibble at the very small degree of attention paid to the constraints of controller bandwidth and power, and the extent t o which high-frequency uncertainty in the plant’ limits the designer’s ability to suppress the effects of plant parameter variation or nonlinearities by high gain. State-space approaches to design occupy the next chapter. At this level, controllability and observability deserve to be located at the end rather than beginning of the chapter, and this is what the authors have chosen to do. Two further chapters contain important material on quantization and sample rate selection. The latter particularly is given with a highly control flavor, considering as it does issues of disturbance rejection and sensitivity to plant parameter variation. Finally, the book has two chapters which introduce the student to two more advanced topics: system identification, a topic of increasing relevance given the ability to hook a computer onto a system, and multivariable and optimal control. The latter chapter develops the earlier state-variable ideas. Somewhat strangely, these chapters are interpolated between the two chapters on quantization and sample rate selection. Topics not dealt with in the book (and this will probably disappoint some instructors) include minimum variance and sel€-tuning regulators. The combination of what is really superlative writing technique, in the broadest sense in which these words can be applied to a text book, and the wise choice of subject matter will guarantee this book a most favorable reception. But readers of the TRANSACTIONS are again warned: it is a textbook on digital control, not digital signal processing.
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01 Jan 1981TL;DR: The purpose of such a system is to distinguish between the wanted signal and unwanted noise, and it can be done according to some optimality criteria, but the success of the processor is highly dependent upon the noise situation, which often is continously varying.
Abstract: During the past two decades extensive work on optimum and adaptive array processing has been carried out. This has been motivated especially by the importance of sonar, seismic processing, radar and the evolution of electronics which have made adaptive systems realizable. The purpose of such a system is to distinguish between the wanted signal and unwanted noise. This processing can be done according to some optimality criteria. The success of the processor is highly dependent upon the noise situation, which often is continously varying. An inflexible processor will be optimum for a certain postulated situation only, and this might never occur. However, if the noise situation is changing slowly so that it can be estimated continously, near optimum processing may be achieved. The amount of improvement for such an adaptive system is a function of the average noise situation. The cost of such a system may direct the choice towards suboptimum solutions.
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01 Jan 1981
TL;DR: In this paper, a detailed history of Submarine Exploration of the Arctic Basin and Contiguous Marginal Sea Ice Zones is described. And the authors present a detailed analysis of the effects of noise sources, radiation, and inhomogeneities in Underwater Sound Propagation.
Abstract: I: Underwater Sound Sources and Propagation.- Noise Sources, Radiation and Mitigation.- Inhomogeneities and Instabilities in Underwater Sound Propagation.- Medium Inhomogeneities and Instabilities: Effects on Spatial and Temporal Processing.- Chemical Sound Absorption in the Sea.- Environmental Tresholds of Ambient Noise in the Ocean.- Underwater Explosives: Scaling of Source Spectra.- The Remote Sensing of Factors Influencing Underwater Acoustics.- Scattering from Inhomogeneities.- Deterministic Propagation Modelling I: Fundamental Principles.- Deterministic Propagation Modelling II: Numerical Results.- Stochastic Propagation Modelling.- A Brief History of Submarine Exploration of the Arctic Basin and Contiguous Marginal Sea Ice Zones.- A Calculation of Complex Wavenumbers of Virtual Modes in a Pekeris Model.- Ambient Noise Generation in Pack Ice.- Acoustic Detection, Communication, and Signal Processing Requirements for the Optional Deployment of SSBNs in the Poseidon-X Mode in Shallow Ocean Waters.- Propagation of Sound in Marine Sediments.- Signal Processing Aspects on Nonlinear Acoustics.- Calculation of the Accurate Difference Frequency Field of a Parametric Circular Piston.- II: Transducer Technology.- Modern Transducers, Theory and Practice.- New Types of Transducers.- Status of Ultrasonic Lens Development.- Control of Radiated Pressure Using State Variable Feedback.- III: Mathematical Fundamentals of Signal Processing.- A Review of Adaptive Antennas.- Adaptive Beamforming with Emphasis on Narrowband Implementation.- Random Acoustic Arrays.- Array Adaptive Noise Cancellation under Signal Leakage Conditions.- Multi Dimensional Constrained Lattice Processor for Adaptive Arrays.- General Detection and Estimation Theory in an Adaptive Context.- Detection and Estimation. A Summary of Results.- Fast Algorithms for Time Domain Adaptive Array Processing.- Time Delay Estimation in a Sensor Array.- Modeling, Algorithmic and Performance Issues in Deepwater Ranging Sound Systems.- Generalized Time Space Approach and Convolution Arrays.- Recent Developments in Least Squares Model Fitting.- Modern Empirical Statistical Spectral Analysis.- Passive Sonar Signal Processing.- Adaptive High Resolution Spatial Discrimination of Passive Sources.- On Array Processing in Non-Stationary Random Fields.- Response of Optimal Beamformer to Broad Angular Spectra.- Two-Dimensional Digital Filtering with Applications to Image Processing.- Two-Dimensional Spectral Estimation by Linear Prediction.- Design of Two-Dimensional Digital Filters for Electron Microscopy.- On the Detection of Fluctuating Signals.- Nonparametric Tests for Coherent Detection.- Estimation of Radial Target Length Using High Resolution Signals.- Eigenvector Directions of Spectral Density Matrix: Applications to Characterization of Sources and Modelling of Propagation Media.- Estimation and Tracking of Frequencies of Sinusoids in Noise.- Dynamic Programming for Phase and Frequency Tracking.- The Underwater Medium as a Generalized Communication Channel.- Physical and Descriptive Approaches to Medium-Induced Acoustic Fluctuation.- Characterisation of Submarine Acoustic Transmission Channels.- IV: Computational Background of Signal Processing.- Application of Micro, Mini and Maxi Processors.- High-Speed Programmable Digital Signal Processing Systems for Underwater Research.- Estimates of the Channel Capacity for Acoustic Underwater Communication.- Application of the Coupling of SAW-TO-DIGITAL Technologies to Fast Underwater Signal Processing.- V: Contributions from Other Fields.- Signal Processing in Geophysics.- Radioastronomy.- The Application of Advanced Pattern Recognition Techniques for the Discrimination between Earthquakes and Nuclear Detonations.- Tomographic Ocean Surveillance.- Summaries of Workshops.- Summaries of Workshops.- List of Chairmen.
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01 Apr 1981TL;DR: MAP performance is compared to that of a conventional processor in dynamic simulated signal and interference environments and advantages of this technique over both conventional and competing adaptive techniques are discussed.
Abstract: The Matched Array Processor (MAP) is quadratic adaptive matrix filter designed to optimize detection and bearing estimation of a desired wideband target signature in the presence of interferences. It is made adaptive in the spatial domain to remove interferences entering through the beam sidelobes and it is spectrally matched to the signal of interest. In this paper, MAP performance is compared to that of a conventional processor in dynamic simulated signal and interference environments. Advantages of this technique over both conventional and competing adaptive techniques is discussed.
01 Aug 1981
TL;DR: In this paper, the eigenvalues of the covariance matrix for an N-element LMS adaptive array are discussed and a simple relation between the array output SINR and one of its eigen values is obtained for the case of strong interference and CW signals.
Abstract: : This report discusses the eigenvalues of the covariance matrix for an N-element LMS adaptive array. The effects of the signal and array parameters on these eigenvalues are discussed. A simple relation between the array output SINR and one of the eigenvalues is obtained for the case of strong interference and CW signals. (Author)