17 research outputs found

    Sensor array signal processing : two decades later

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    Caption title.Includes bibliographical references (p. 55-65).Supported by Army Research Office. DAAL03-92-G-115 Supported by the Air Force Office of Scientific Research. F49620-92-J-2002 Supported by the National Science Foundation. MIP-9015281 Supported by the ONR. N00014-91-J-1967 Supported by the AFOSR. F49620-93-1-0102Hamid Krim, Mats Viberg

    Array and multichannel signal processing using nonparametric statistics

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    In array signal processing a group of sensors located at distinct spatial locations is deployed to measure a propagating wavefield. The multichannel output is then processed to provide information about parameters of interest. Application areas include smart antennas in communications, radar, sonar and biomedicine. When deriving array signal processing algorithms the noise is typically modeled as a white Gaussian random process. A shortcoming of the estimation procedures derived under Gaussian assumption is that they are extremely sensitive to deviations from the assumed model, i.e. they are not robust. In real-world applications the assumption of white Gaussian noise is not always valid. Consequently, there has been a growing interest in estimation methods which work reliably in both Gaussian and non-Gaussian noise. In this thesis, new statistical procedures for array and multichannel signal processing are developed. In the area of array signal processing, the work concentrates on high-resolution subspace-based Direction Of Arrival (DOA) estimation and estimation of the number of source signals. Robust methods for DOA estimation and estimation of the number of source signals are derived. Spatial-smoothing based extensions of the techniques to deal with coherent signals are also derived. The methods developed are based on multivariate nonparametric statistics, in particular sign and rank covariance matrices. It is shown that these statistics may be used to obtain convergent estimates of the signal and noise subspaces for a large family of symmetric noise distributions. Simulations reveal that the techniques developed exhibit near-optimal performance when the noise distribution is Gaussian and are highly reliable if the noise is non-Gaussian. Multivariate nonparametric statistics are also applied to frequency estimation and estimation of the eigenvectors of the covariance matrix. Theoretical justification for the techniques is shown and their robust performance is illustrated in simulations.reviewe

    High-resolution sonar DF system

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    One of the fundamental problems of sonar systems is the determination of the bearings of underwater sources/targets. The classical solution to this problem, the 'Conventional Beamformer', uses the outputs from the individual sensors of an acoustic array to form a beam which is swept across the search sector. The resolution of this method is limited by the beam width and narrowing this beam to enhance the resolution may have some practical problems, especially in low frequency sonar, because of the physical size of the array needed. During the past two decades an enormous amount of work has been done to develop new algorithms for resolution enhancements beyond that of the Conventional Beamformer. However, most of these methods have been based on computer simulations and very little has been published on the practical implementation of these algorithms. One of the main reasons for this has been the lack of hardware that can handle the relatively heavy computational load of these algorithms. However, there have been great advances in semiconductor and computer technologies in the last few years which have led to the availability of more powerful computational and storage devices. These devices have opened the door to the possibility of implementing these high-resolution Direction Finding (DF) algorithms in real sonar systems. The work presented in this thesis describes a practical implementation of some of the high-resolution DF algorithms in a simple sonar system that has been designed and built for this purpose. [Continues.

    Signal Subspace Processing in the Beam Space of a True Time Delay Beamformer Bank

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    A number of techniques for Radio Frequency (RF) source location for wide bandwidth signals have been described that utilize coherent signal subspace processing, but often suffer from limitations such as the requirement for preliminary source location estimation, the need to apply the technique iteratively, computational expense or others. This dissertation examines a method that performs subspace processing of the data from a bank of true time delay beamformers. The spatial diversity of the beamformer bank alleviates the need for a preliminary estimate while simultaneously reducing the dimensionality of subsequent signal subspace processing resulting in computational efficiency. The pointing direction of the true time delay beams is independent of frequency, which results in a mapping from element space to beam space that is wide bandwidth in nature. This dissertation reviews previous methods, introduces the present method, presents simulation results that demonstrate the assertions, discusses an analysis of performance in relation to the Cramer-Rao Lower Bound (CRLB) with various levels of noise in the system, and discusses computational efficiency. One limitation of the method is that in practice it may be appropriate for systems that can tolerate a limited field of view. The application of Electronic Intelligence is one such application. This application is discussed as one that is appropriate for a method exhibiting high resolution of very wide bandwidth closely spaced sources and often does not require a wide field of view. In relation to system applications, this dissertation also discusses practical employment of the novel method in terms of antenna elements, arrays, platforms, engagement geometries, and other parameters. The true time delay beam space method is shown through modeling and simulation to be capable of resolving closely spaced very wideband sources over a relevant field of view in a single algorithmic pass, requiring no course preliminary estimation, and exhibiting low computational expense superior to many previous wideband coherent integration techniques

    Signal Subspace Processing in the Beam Space of a True Time Delay Beamformer Bank

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    A number of techniques for Radio Frequency (RF) source location for wide bandwidth signals have been described that utilize coherent signal subspace processing, but often suffer from limitations such as the requirement for preliminary source location estimation, the need to apply the technique iteratively, computational expense or others. This dissertation examines a method that performs subspace processing of the data from a bank of true time delay beamformers. The spatial diversity of the beamformer bank alleviates the need for a preliminary estimate while simultaneously reducing the dimensionality of subsequent signal subspace processing resulting in computational efficiency. The pointing direction of the true time delay beams is independent of frequency, which results in a mapping from element space to beam space that is wide bandwidth in nature. This dissertation reviews previous methods, introduces the present method, presents simulation results that demonstrate the assertions, discusses an analysis of performance in relation to the Cramer-Rao Lower Bound (CRLB) with various levels of noise in the system, and discusses computational efficiency. One limitation of the method is that in practice it may be appropriate for systems that can tolerate a limited field of view. The application of Electronic Intelligence is one such application. This application is discussed as one that is appropriate for a method exhibiting high resolution of very wide bandwidth closely spaced sources and often does not require a wide field of view. In relation to system applications, this dissertation also discusses practical employment of the novel method in terms of antenna elements, arrays, platforms, engagement geometries, and other parameters. The true time delay beam space method is shown through modeling and simulation to be capable of resolving closely spaced very wideband sources over a relevant field of view in a single algorithmic pass, requiring no course preliminary estimation, and exhibiting low computational expense superior to many previous wideband coherent integration techniques

    Efficient Beamspace Eigen-Based Direction of Arrival Estimation schemes

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    The Multiple SIgnal Classification (MUSIC) algorithm developed in the late 70\u27s was the first vector subspace approach used to accurately determine the arrival angles of signal wavefronts impinging upon an array of sensors. As facilitated by the geometry associated with the common uniform linear array of sensors, a root-based formulation was developed to replace the computationally intensive spectral search process and was found to offer an enhanced resolution capability in the presence of two closely-spaced signals. Operation in beamspace, where sectors of space are individually probed via a pre-processor operating on the sensor data, was found to offer both a performance benefit and a reduced computationa1 complexi ty resulting from the reduced data dimension associated with beamspace processing. Little progress, however, has been made in the development of a computationally efficient Root-MUSIC algorithm in a beamspace setting. Two approaches of efficiently arriving at a Root-MUSIC formulation in beamspace are developed and analyzed in this Thesis. In the first approach, a structura1 constraint is placed on the beamforming vectors that can be exploited to yield a reduced order polynomial whose roots provide information on the signal arrival angles. The second approach is considerably more general, and hence, applicable to any vector subspace angle estimation algorithm. In this approach, classical multirate digital signal processing is applied to effectively reduce the dimension of the vectors that span the signal subspace, leading to an efficient beamspace Root-MUSIC (or ESPRIT) algorithm. An auxiliaay, yet important, observation is shown to allow a real-valued eigenanalysis of the beamspace sample covariance matrix to provide a computational savings as well as a performance benefit, particularly in the case of correlated signal scenes. A rigorous theoretical analysis, based upon derived large-sample statistics of the signal subspace eigenvectors, is included to provide insight into the operation of the two algorithmic methodologies employing the real-valued processing enhancement. Numerous simulations are presented to validate the theoretical angle bias and variance expressions as well as to assess the merit of the two beamspace approaches

    Combined-channel instantaneous frequency analysis for audio source separation based on comodulation

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    Thesis (Ph. D.)--Harvard-MIT Division of Health Sciences and Technology, 2008.Includes bibliographical references (p. 295-303).Normal human listeners have a remarkable ability to focus on a single sound or speaker of interest and to block out competing sound sources. Individuals with hearing impairments, on the other hand, often experience great difficulty in noisy environments. The goal of our research is to develop novel signal processing methods inspired by neural auditory processing that can improve current speech separation systems. These could potentially be of use as assistive devices for the hearing impaired, and in many other communications applications. Our focus is the monaural case where spatial information is not available. Much perceptual evidence indicates that detecting common amplitude and frequency variation in acoustic signals plays an important role in the separation process. The physical mechanisms of sound generation in many sources cause common onsets/offsets and correlated increases/decreases in both amplitude and frequency among the spectral components of an individual source, which can potentially serve as a distinct signature. However, harnessing these common modulation patterns is difficult because when spectral components of competing sources overlap within the bandwidth of a single auditory filter, the modulation envelope of the resultant waveform resembles that of neither source. To overcome this, for the coherent, constant-frequency AM case, we derive a set of matrix equations which describes the mixture, and we prove that there exists a unique factorization under certain constraints. These constraints provide insight into the importance of onset cues in source separation. We develop algorithms for solving the system in those cases in which a unique solution exists. This work has direct bearing on the general theory of non-negative matrix factorization which has recently been applied to various problems in biology and learning. For the general, incoherent, AM and FM case, the situation is far more complex because constructive and destructive interference between sources causes amplitude fluctuations within channels that obscures the modulation patterns of individual sources.(cont.) Motivated by the importance of temporal processing in the auditory system, and specifically, the use of extrema, we explore novel methods for estimating instantaneous amplitude, frequency, and phase of mixtures of sinusoids by comparing the location of local maxima of waveforms from various frequency channels. By using an overlapping exponential filter bank model with properties resembling the cochlea, and combining information from multiple frequency bands, we are able to achieve extremely high frequency and time resolution. This allows us to isolate and track the behavior of individual spectral components which can be compared and grouped with others of like type. Our work includes both computational and analytic approaches to the general problem. Two suites of tests were performed. The first were comparative evaluations of three filter-bank-based algorithms on sets of harmonic-like signals with constant frequencies. One of these algorithms was selected for further performance tests on more complex waveforms, including AM and FM signals of various types, harmonic sets in noise, and actual recordings of male and female speakers, both individual and mixed. For the frequency-varying case, initial results of signal analysis with our methods appear to resolve individual sidebands of single harmonics on short time scales, and raise interesting conceptual questions on how to define, use and interpret the concept of instantaneous frequency. Based on our results, we revisit a number of questions in current auditory research, including the need for both rate and place coding, the asymmetrical shapes of auditory filters, and a possible explanation for the deficit of the hearing impaired in noise.by Barry David Jacobson.Ph.D

    Linear Predictive Spectral Analysis via the Lp Norm

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    This study involves linear predictive spectral analysis under the general LP norm; both one dimensional and two dimensional spectral estimation algorithms are developed. The objective in this study is determination of frequency resolution capability for various LP normed solutions to linear predictive spectral estimation equations. A modified residual steepest descent algorithm is utilized to generate the required solution. The research presented in this thesis could not have been accomplished without the support of the Oklahoma State University Research Consortium For Well Log Data Enhancement Via Signal Processing. The member companies of this consortium include Amococ Production Company, Area Oil and Gas Company, Cities Service Oil and Gas Corporation, Conoco, Exxon, IBM, Mobil Research and Development, Phillips Petroleum Corporation, Sohio Petroleum Company, and Texaco.Electrical Engineerin

    Angle-of-Arrival Measurement Techniques for Enhanced Positioning in Beyond 5G Systems

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    The new generation of mobile communication systems introduces new methods and technologies that may enhance positioning accuracy in some scenarios when the GNSS system cannot meet the requirements, such as indoor positioning and outdoor autonomous driving. The 3GPP standard and for the first time included the angle measurement as new positioning methods in 5G. The Angle of Arrival (AoA) is the angle measurement method on the uplink direction that can enjoy the new capabilities in 5G systems to enhance the positioning downs to centimeters.The new generation of mobile communication systems introduces new methods and technologies that may enhance positioning accuracy in some scenarios when the GNSS system cannot meet the requirements, such as indoor positioning and outdoor autonomous driving. The 3GPP standard and for the first time included the angle measurement as new positioning methods in 5G. The Angle of Arrival (AoA) is the angle measurement method on the uplink direction that can enjoy the new capabilities in 5G systems to enhance the positioning downs to centimeters. Multiple Signal Classification Method (MUSIC) is a high-accuracy super-resolution algorithm for AoA estimation. The MUSIC method for estimating AoA has many shortcomings that make it unsuitable for a wide variety of scenarios. Correlated multipath signals substantially reduce estimation accuracy. Additionally, this method is a searching algorithm that requires a significant amount of time to resolve AoA. In this thesis, a CASCADE algorithm was proposed to overcome MUSIC's constraints by estimating a coarse range of AoA using a rapid AoA algorithm and then passing that range to the second stage represented by MUSIC to estimate AoA correctly. Multipath signals were eliminated by modifying the proposed CASCADE to detect only the line of sight (LOS), which is the essential path for angular localization. Additionally, the thesis compares many AoA algorithms in the context of 5G systems. A sounding reference signal (SRS) in the mm-wave band was generated according to the 3GPP standards and utilized as the input to those algorithms. A simulation was conducted throughout this thesis by evaluating six AoA algorithms: Bartlet Beamforming, MVDR, MUSIC, ESPRIT, FFT, and the proposed CASCADE method. The results showed that the proposed algorithm achieves the best performance when using less than 64 array antenna elements. On the other hand, FFT alone can provide high accuracy when using an ultra massive antenna system (e.g., 256,512,1024). Additionally, the findings observed the effect of key parameters on the performance of AoA algorithms, such as low SNR, a small number of snapshots (samples), and the effect of multipath signals

    Acoustical measurements on stages of nine U.S. concert halls

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