Improved time-delay estimates of underwater acoustic signals usingbeamforming and prefiltering techniques

Passive sonar systems that localize broadband sources of acoustic energy estimate the difference in arrival times (or time delays) of an acoustic wavefront at spatially separated hydrophones, The output amplitudes from a given pair of hydrophones are cross-correlated, and an estimate of the time delay is given by the time lag that maximizes the cross correlation function. Often the time-delay estimates are corrupted by the presence of noise. By replacing each of the omnidirectional hydrophones with an array of hydrophones, and then cross-correlating the beamformed outputs of the arrays, the author shows that the effect of noise on the time-delay estimation process is reduced greatly. Both conventional and adaptive beamforming methods are implemented in the frequency domain and the advantages of array beamforming (prior to cross-correlation) are highlighted using both simulated and real noise-field data. Further improvement in the performance of the broadband cross-correlation processor occurs when various prefiltering algorithms are invoked     

Spatial processing of signals received by platform mounted sonar

Acoustic signals received by platform mounted sonar arrays can be spatially processed to enhance the detection of targets in the presence of both ambient and platform generated (self) noise. Ambient noise in the ocean, such as that due to distant shipping or biological choruses, are known to be spatially correlated. The platform generated noise will be of near-field origin and may not be received by all elements in the array. In this paper we investigate the performance of the minimum variance distortionless response (MVDR) beamformer and the recently introduced Fourier integral method (FIM) and compare their performances with the conventional beamformer. Real passive sonar data, obtained from a platform mounted sparse linear array of hydrophones, is used to study the performance of the beamformers in a typical sonar environment. It is shown that in the absence of self noise, when the array is accurately calibrated the MVDR beamformer will perform very well, but when sensor gain or phase errors are present the performance of the MVDR beamformer is degraded. Further, the MVDR beamformer is unable to reject the self noise which is not "seen" by the entire array. FIM however seems to perform well and a modified version of FIM, which we call weighted FIM (WFIM), is shown to perform better and is at worst comparable to a well calibrated MVDR beamformer     

Remedying the effects of array shape distortion on the spatialfiltering of acoustic data from a line array of hydrophones

The effects of both small perturbations and large deformations to the array's shape on both conventional and adaptive beamformers are shown for two frequencies: the spatial Nyquist frequency (or design frequency) of the array and a frequency about three times greater. Large shape deformations lead to a decrease in the conventional beamformer's output power for a beam steered in the direction of the signal source, together with an increase in the sidelobe levels (or secondary maxima), while small perturbations in the array shape have little effect. Signal suppression is observed to be far greater for the adaptive beamformer because it is very sensitive to system errors. The imposition of a weight norm constraint on the adaptive beamformer reduces the signal suppression only for small shape perturbations array shape estimation techniques are needed to reduce signal suppression for large shape deformations. The adverse effects of a nonlinear array shape on both conventional and adaptive beamforming are shown to be substantially reduced by applying techniques that estimate the coordinates of the hydrophones prior to beamforming     

Adaptive port-starboard beamforming of triplet sonar arrays

For a low-frequency active sonar (LFAS) with a triplet receiver array, it is not clear in advance which signal processing techniques optimize its performance. Here, several advanced beamformers are analyzed theoretically, and the results are compared to experimental data obtained in sea trials. Triplet arrays are single line arrays with three hydrophones on a circular section of the array. The triplet structure provides the ability to solve the notorious port-starboard (PS) ambiguity problem of ordinary single-array receivers. More importantly, the PS rejection can be so strong that it allows to unmask targets in the presence of strong coastal reverberation or traffic noise. The theoretical and experimental performance of triplet array beamformers is determined in terms of two performance indicators: array gain and PS rejection. Results are obtained under several typical acoustic environments: sea noise, flow noise, coastal reverberation, and mixtures of these. A new algorithm for (beam space) adaptive triplet beamforming is implemented and tuned. Its results are compared to those of other triplet beamforming techniques (optimum and cardioid beamforming). These beamformers optimize for only one performance indicator, whereas in theory, the adaptive beamformer gives the best overall performance (in any given environment). The different beamformers are applied to data obtained with an LFAS at sea. Analysis shows that adaptive triplet beamforming outperforms conventional beamforming algorithms. Adaptive triplet beamforming provides strong PS rejection, allowing the unmasking of targets in the presence of strong directional reverberation (e.g., from a coast) and at the same time provides positive array gain in most environments.     

基于水听器阵的广义相关波束形成算法研究

传统上都是使用基于常规波束形成(CBF)输出功率的不同进行声源的被动检测和定位。广义相关波束形成,是一种旨在将平均Toep litz相关函数用于估计波束形成的输出功率。从线性矢量水听器阵出发,分析了常规波束形成,对广义相关波束形成(GCBF)进行了深入研究,通过对广义相关波束形成器进行“W ilson”和“Bartlett”加权,可以分别得到FIM(WFIM)和CBF,对这三种波束形成器进行了理论分析和计算机仿真,结果表明三个波束形成器的分辨能力从高到低依次是:FIM、WFIM、CBF。     

Adaptive beamforming of a towed array during a turn

During maneuvering, towed array beamforming degrades if a straight array is assumed. This is especially true for high-resolution adaptive beamforming. It is experimentally demonstrated that adaptive beamforming is feasible on a turning array, provided that array shape is estimated. The array shape can be inferred solely from the coordinates of the tow vessel's Global Positioning System (GPS) without any instrumentation in the array. Based on estimated array shape from the GPS, both the conventional beamformer and the white noise constrained (WNC) adaptive beamformer are shown to track the source well during a turn. When calculating the weight vector in the WNC approach, a matrix inversion of the cross-spectral density matrix is involved. This matrix inversion can be stabilized by averaging the cross-spectral density matrix over neighboring frequencies. The proposed algorithms have been tested on real data with the tow-vessel making 45/spl deg/ turns with a 500-m curvature radius. While turning, the improvement in performance over the assumption of a straight array geometry was up to 5 dB for the conventional beamformer and considerably larger for the WNC adaptive beamformer.     

Optimization of conventional beamformer shading weights for conformal velocity sonar

A numerical optimization technique that uses sonar array noise measurements is used to determine conventional shading weights that maximize the broadband deflection coefficient at the output of the optimal square-law detector, across a frequency band of interest. This process maintains the structure of the conventional processor while providing performance improvement typical of adaptive techniques. The performance of the optimized time domain delay-and-sum beamformer is compared with that of the traditional beamformer that uses conventionally chosen shading weights. Application of this method to conformal velocity sonar array data is shown to provide large improvements in performance over heuristic designs.     

基于高阶累积量的自适应波束形成

常规波束形成器仅限于利用信号的二阶统计特性。作者采用高阶累积量估计期望信号的导向矢量 (Steering Vector) ,实现了一种基于高阶累积量的自适应波束形成器 (HCAB)。该波束形成器利用信号的更高阶统计特性 ,减少了对阵列流型的依赖 ,具有较好的容差性 ,能自动跟踪信号。数值模拟实验表明该波束形成器工作良好。     

Limitations on the overlap-correlator method imposed by noise andsignal characteristics

Limitations on the performance of the overlap-correlator method of forming a passive synthetic aperture are derived. The technique uses the overlap of the array in sequential positions to estimate a series of phase correction factors that compensate for the motion of the array over time. It is of primary interest to optimize this overlap with respect to the effects of random noise. By minimizing the variance of the estimates of the set of phase correction factors, it is found that the optimal overlap is one-half the length of the physical array. Using this optimal overlap, the bounds on the usable spatial response are then determined as a function of signal-to-noise ratio and the number of hydrophones in the physical array. The ability of the overlap-correlator algorithm to synthesize a coherent aperture is investigated for the case of multiple sources in the absence of noise     

Performance comparison of high resolution bearing estimationalgorithms using simulated and sea test data

The performance of both the Capon and the MUSIC high resolution bearing estimation algorithms is investigated using both simulated data and sea test data collected with an experimental planar array. The major problem with these estimators is their sensitivity to both system errors and deviations from the assumed noise model. To alleviate this problem, two methods for preprocessing the data before they are input into the high-resolution algorithm are investigated: beam space and sector focused stability. The performance of both high-resolution estimators is examined, using both types of preprocessing, and the results are compared with those for the standard element-space (ES) techniques, assuming both finite cross-spectral-matrix (CSM) averaging errors and weakening target strengths. For the Capon estimator the performance is only superior to the standard element space technique when the CSM is calculated using a small number of averages. For the MUSIC estimator, both preprocessing techniques give clearly superior results over standard space techniques, with the SFS preprocessor performing the best     

Adaptive beamforming: spatial filter designed blocking matrix

Controlling the resolution in adaptive beamformers is often crucial. A simple method that works for both narrow-band and broad-band arrays is presented. This method is based on the normalized leaky LMS algorithm in conjunction with a generalized sidelobe canceller (GSC) structure, where the GSC is designed using a spatial filtering approach. In essence, the suppression of the spatial filters and the implicit noise of the leaky LMS algorithm together determine the adaptive beamformer. Analytical expressions are given for the Wiener filters and the output spectrum versus frequency and point source location. These expressions are employed in the design specification of the spatial filters and to obtain conditions for a controlled quiescent beamformer response. Simulation results are presented to illustrate the behavior of the array     

Particle-Velocity-Field Difference Smoothing for Coherent Source Localization in Spatially Nonuniform Noise

This communication considers the problem of estimating 2-D directions of arrival (DOAs) of multiple coherent signals under spatially nonuniform noise (spatially inhomogeneous temporary white noise) using an array of vector hydrophones. A novel preprocessing method called particle-velocity-field difference smoothing (PVFDS) is proposed. The key idea underlying the PVFDS is to remove the spatially nonuniform noise by using the matrix difference of pairs of particle-velocity data correlation matrices, and to decorrelate the coherent signals by summing these difference correlation matrices. Unlike most of other existing preprocessing techniques, such as spatial smoothing and forward–backward averaging, the PVFDS processing does not decrease the array aperture. For arbitrary array geometries, the PVFDS can resolve up to four coherent signals, and for centro–symmetric arrays, forward–backward averaging can double this number to eight. Monte Carlo simulations illustrate that the PVFDS-based eigenstructure algorithms can offer better performance than the particle-velocity-field smoothing (PVFS)-based counterparts.      

舷侧阵CBF和OBF的阵增益比较

以舷侧阵作为研究对象,首先介绍了常规波束形成器(CBF)和最佳波束形成器(OBF)的波束输出信号模型,然后推导了信号+噪声场合下的阵增益表达式,比较了在不同信号频率和不同信号源方向角下,CBF和OBF的阵增益,最后在增加了有指向性的CW干扰后,分析了此场合下的阵增益,并比较了不同的信号频率下,干扰源方向角的变化对两类波束形成器阵增益的影响。     

Extended-aperture underwater acoustic multisource azimuth/elevationdirection-finding using uniformly but sparsely spaced vector hydrophones

Aperture extension is achieved in this novel ESPRIT-based two-dimensional angle estimation scheme using a uniform rectangular array of vector hydrophones spaced much farther apart than a half-wavelength. A vector hydrophone comprises two or three spatially co-located, orthogonally oriented identical velocity hydrophones (each of which measures one Cartesian component of the underwater acoustical particle velocity vector-field) plus an optional pressure hydrophone. Each incident source's directions-of-arrival are determined from the source's acoustical particle velocity components, which are extracted by decoupling the data covariance matrix's signal-subspace eigenvectors using the lower dimensional eigenvectors obtainable by ESPRIT. These direction-cosine estimates are unambiguous but have high variance; they are used as coarse references to disambiguate the cyclic phase ambiguities in ESPRIT's eigenvalues when the intervector-hydrophone spacing exceeds a half-wavelength. In one simulation scenario, the estimation standard deviation decreases with increasing intervector-hydrophone spacing up to 12 wavelengths, effecting a 97% reduction in the estimation standard deviation relative to the half-wavelength case. This proposed scheme and the attendant vector-hydrophone array outperform a uniform half-wavelength spaced pressure-hydrophone array with the same aperture and slightly greater number of component hydrophones by an order of magnitude in estimation standard deviation. Other simulations demonstrate how this proposed method improves underwater acoustic communications link performance. The virtual array interpolation technique would allow this proposed algorithm to be used with irregular array geometries     

ROC evaluation of adaptive beamforming in a simulated shallow waterenvironment

An optimal evaluation of adaptive beamforming techniques in a reverberation-limited shallow water environment is presented. A comprehensive simulation, using the sonar simulation toolset (SST) software in conjunction with the generic sonar model (GSRT) software, is used to create realistic beam data complete with target, noise, and reverberation. Adaptive beamforming techniques from the recursive least squares (RLS) family are applied to enhance detection performance via interference rejection. Two techniques are considered: linearly constrained beamforming using the minimum variance distortionless response (MVOR) beamformer and constrained adaptive noise cancelling (ANC) using the joint process least squares lattice (JPLSL) algorithm. Target detection trials, summarized in the form of receiver operator characteristics (ROC), are used to evaluate performance of the two adaptive beamformers. Results demonstrate mixed performance in reverberation-limited shallow water environments     

Theory and analytical performance evaluation of generalizedcorrelation beamformers

Traditionally, passive detection and localization of an acoustic source has been based on exploiting the relative differences in temporally averaged power outputs of contiguous beams of an element-weighted beamformer. An alternate approach, the generalized correlation beamformer (GCBF), is proposed where a weighted Toeplitz-averaged (spatially averaged) correlation function is used to estimate the beamformer output power. All element-weight sequences can be transformed into correlation-weight sequences through a convolution operation. Additional weight sequences which cannot be generated from a convolution of real element-weight sequences are available for use in the GCBF. A special case of the GCBF was proposed by Wilson et al. (1995) in which the correlation-weights are set to unity, a correlation-weight sequence which cannot be obtained from any classical element-weight sequence. Although such a “boxcar” correlation-weight sequence produces a sharper main peak power response (improved resolution), it has the undesirable effect of producing abnormally high (positive and negative power) sidelobes. General analytical performance bounds are developed that accurately reflect the GCBF detection and bearing localization performance for a noise model that includes spatially white noise and spatially discrete interferers (clutter). Analysis results indicate that the GCBF with Bartlett correlation-weights outperformed the GCBF with unity correlation-weights for both detection and bearing estimation except when the clutter bearing is close to the signal bearing     

Range localization of 10-100 km explosions by means of an endfire array and a waveguide Invariant

Short acoustical signals like those caused by explosions will in a waveguide split into mode arrivals. If the distance is long enough, they can at the receiver be resolved in time with appropriate narrowband filters. They can simultaneously be resolved in vertical angle (incidence-) with an endfire array and a beamformer. Combined in a beam-time diagram the arrivals will line up along a straight line. The slope of this line is invariant with frequency, mode indexes, source and receiver depths. It can conveniently be linked to the so-called waveguide invariant /spl beta/. An alternative approach to /spl beta/ is to compute it from the bathymetric profile. This is valid for range variable waveguides under adiabatic conditions, constant water sound speed over a harder bottom, and small grazing angles. Together these two approaches to /spl beta/ can be combined in a formula, where direct range determination is the end product. The applicability of the method is demonstrated on data from an experiment at sea. An 820-m array with 10 hydrophones was deployed at the bottom in 320-m water depth. For two endfire runs in opposite directions, small explosive charges out to 115 km were used as sound sources. Typical range estimation errors were 5-10%.     

On Performance of Array Signal Processing

A unified treatment for performance evaluation of various array signal processors is presented. Detection performance is expressed in terms of the parameter of the power-type receiver operating characteristic (ROC) for optimum, beamformer, and null-steerer detectors. Estimation performance is analyzed in terms of the normalized mean-square error (MSE) for minimum mean-square error (MMSE) and maximum likelihood estimators (MLE's) under a varying noise environment. Sensitivity of the detection/estimation performance to the varying internal and directional noise sources is investigated. An interesting inverse relationship is presented between the normalized MSE of the MMSE estimator and the power-type ROC parameter for the optimum detector.     

Calculation of the shape of a towed underwater acoustic array

An important area of towed underwater acoustic research is the determination of the 3D positions of all hydrophones in the array. Although there are a number of methods available that provide position information at a small number of locations along the array, an interpolation scheme is needed that will permit the estimation of the position of all hydrophones so that further processing of acoustic data may proceed. An interpolation technique based on a twisted quartic spline approximation to a space curve is presented. This technique provides the advantages of numerical stability, necessary smoothness, and satisfaction of physical boundary conditions. Most importantly, it permits the estimation of the positions of all hydrophones in an array     

Comparison of Kalman and least squares filters for locatingautonomous very low frequency acoustic sensors

The Marine Physical Laboratory has designed, fabricated, and taken to sea self-contained, freely drifting acoustic sensors which can measure signal propagation and ambient ocean noise in the 1-20-Hz band for up to 25-hour periods. The deployment of several freely drifting floats forms an array of sensors whose outputs can be combined after the experiment with a beamformer. A Kalman filter and a least-squares estimator have been developed to estimate float positions from travel-time measurements. Computer simulation is used to compare filter performance-under several deployment scenarios. Results show that the Kalman filter performs better than the least-squares filter when the floats are subjected to small-magnitude accelerations between measurements. Neither filter was sensitive to relatively major changes in deployment geometry as long as the sound-speed profile is known exactly