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.     

Transiting aircraft parameter estimation using underwater acousticsensor data

Sound from an airborne source travels to a receiver beneath the sea surface via a geometric path that is most simply described using ray theory, where the atmosphere and the sea are assumed to be isospeed sound propagation media separated by a planar surface (the air-sea interface). This theoretical approach leads to the development of a time-frequency model for the signal received by a single underwater acoustic sensor and a time-delay model for the signals received by a pair of spatially separated underwater acoustic sensors. The validity of these models is verified using spatially averaged experimental data recorded from a linear array of hydrophones during various transits of a turboprop aircraft. The same approach is used to solve the inverse time-frequency problem, that is, estimation of the aircraft's speed, altitude, and propeller blade rate given the observed variation with time of the instantaneous frequency of the received signal. Similarly, the inverse time-delay problem is considered whereby the speed and altitude of the aircraft are estimated using the differential time-of-arrival information from each of two adjacent pairs of widely spaced hydrophones (with one hydrophone being common to each pair). It is found that the solutions to each of the inverse problems provide reliable estimates of the speed and altitude of the aircraft, with the inverse time-frequency method also providing an estimate that closely matches the actual propeller blade rate     

Evaluation of the dominant mode rejection beamformer using reducedintegration times

Increasing the number of hydrophones in an array should increase beamformer performance. However, when the number of hydrophones is large, integration times must be long enough to give accurate cross-spectral matrix (CSM) estimates, but short enough so that the dynamic behavior of the noise described by the CSM is captured. The dominant mode rejection (DMR) beamformer calculates adaptive weights based on a reduced rank CSM estimate, where the CSM estimate is formed with a subset of the largest eigenvalues and their eigenvectors. Since the largest eigenvalue/eigenvector pairs are estimated rapidly, the integration time required is reduced. The purpose of this study was to examine the DMR beamformer performance using a bottom-mounted horizontal line array in a shallow-water environment. The data were processed with a fully adaptive beamformer and the DMR beamformer. The DMR beamformer showed better performance than the fully adaptive beamformer when using arrays with larger numbers of hydrophones. Thus, in highly dynamic noise environments, the DMR beamformer may be a more appropriate implementation to use for passive sonar detection systems     

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     

Angular-spectral decomposition beamforming for acoustic arrays

The method of principal component beamforming described in this paper is an array data reduction method that allows one to observe the statistically uncorrelated components of wave energy arriving at an array of acoustic sensors. The method can be used to process array data so as to observe and identify the sources of noise, both environmental and self noise. After identifying the sources of noise, the method of principal components can be used to discriminate signal from noise. The method can be applied to active systems (subbottom profilers) as well as passive systems. A model of isotropic noise and incident bandlimited plane waves is used to study array resolution and bandwidth effects. Experimental data from a2 times 3planar acoustic array were used to identify sources of hydro-flow related noise in an underwater vehicle. In all cases studied, the technique provides a maximum spatial information analysis method to the observer.     

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     

True time-delay bandpass beamforming

A simple digital scheme for bandpass time-domain beamforming that is applicable when there is significant change in the envelope of the signal across the extent of an array of hydrophones is presented. The three requisite processes to obtain complex-baseband beam signals, namely, down conversion to complex baseband, time delay of the complex envelope, and phase rotation, are discussed. Down conversion is accomplished by subsampling the bandpass signal and an efficient interpolation technique is employed for time registration of these samples. Since the same interpolation technique is u sed to implement the time delay, the two processes are combined. With further modification to effect the phase rotation, it is shown how the three requisite processes for bandpass beamforming can be accomplished with a single analog-to-digital converter and two short finite impulse response filters     

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     

基于虚拟仪器的水下目标二维角估计

利用LabVIEW软件,通过8通道数据采集卡和均匀圆阵对水下目标的噪声进行采集和处理。结合一维直线阵波束形成理论,实现了对水下目标二维方向角估计实验研究和算法验证。实验证明利用虚拟仪器方便地实现了对水声信号的采集、处理,以及在方位估计时,为传感器布阵和算法的确定提供参考。     

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。     

An environmental assessment in the Strait of Sicily: measurementand analysis techniques for determining bottom and oceanographicproperties

In October 1997, the EnVerse 97 shallow-water acoustic experiments were jointly conducted by SACLANT Centre, TNO-FEL, and DERA off the coast of Sicily, Italy. The primary goal of the experiments was to determine the sea-bed properties through inversion of acoustic data. Using a towed source, the inversion method is tested at different source/receiver separations in an area with a range-dependent bottom. The sources transmitted over a broadband of frequencies (90-600 Hz) and the signals were measured on a vertical array of hydrophones. The acoustic data were continuously collected as the range between the source and receiving array varied from 0.5 to 6 km. An extensive seismic survey was conducted along the track providing supporting information about the layered structure of the bottom as well as layer compressional sound speeds. The oceanic conditions were assessed using current meters, satellite remote sensing, wave height measurements, and casts for determining conductivity and temperature as a function of water depth. Geoacoustic inversion results taken at different source/receiver ranges show sea-bed properties consistent with the range-dependent features observed in the seismic survey data. These results indicate that shallow-water bottom properties may be estimated over large areas using a towed source fixed receiver configuration     

Methods of array element localization for a towed underwateracoustic array

The location of the hydrophones on a towed underwater acoustic array as a function of time (array element localization) is needed for signal processing. Methods to perform this localization using least squares polynomial fitting to data from depth sensors, heading sensors, and sensors detecting a ping from a single source are discussed. Arc distance along the array is used as the independent parameter so that all solutions are constrained to be space curves. Examples of application to real data are presented, and techniques to discriminate against bad sensor data are discussed     

Application of beam forming techniques to measurements of acoustic scattering from the ocean bottom

Signals from an explosive source backscattered from the seafloor and received at long range by hydrophones of a towed array are processed to estimate the directional distribution of energy for a given time increment. As assembly of these data shows the time and amplitude of scattering features, and after conversion to distance, the geographic location of the return. A frequency-domain beam-forming procedure is used in which beam levels are averaged over a given band of a broad-band source. The processing is applied to experimental data obtained in the southern Tyrrhenian Sea. The major backscattering occurred at the Baconi Seamounts and the coastal margin of Sardinia.     

Acoustic vector-sensor correlations in ambient noise

Most array-processing methods require knowledge of the correlation structure of the noise. While such information may sometimes be obtained from measurements made when no sources are present, this may not always be possible. Furthermore, measurements made in-situ can hardly be used to analyze system performance before deployment. The development of models of the correlation structure under various environmental assumptions is therefore very important. In this paper, we obtain integral and closed form expressions for the auto- and cross-correlations between the components of an acoustic vector sensor (AVS) for a wideband-noise field, under the following assumptions concerning its spatial distribution: 1) azimuthal independence; 2) azimuthal independence and elevational symmetry; and 3) spherical isotropy. We also derive expressions for the cross-covariances between all components of two spatially displaced AVSs in a narrowband-noise field under the same assumptions. These results can be used to determine the noise-covariance matrix of an array of acoustic vector sensors in ambient noise. We apply them to a uniform linear AVS array to asses its beamforming capabilities and localization accuracy, via the Cramer-Rao bound, in isotropic and anisotropic noise     

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     

A Study of Spatial Processing Gain in Underwater Acoustic Communications

Spatial processing, including beamforming and diversity combining, is widely used in communications to mitigate intersymbol interference (ISI) and signal fading caused by multipath propagation. Beamforming suppresses ISI (and noise) by eliminating multipath (and noise) arrivals outside the signal beam. Beamforming requires the signals to be highly coherent between the receivers. Diversity combining combats ISI as well as signal fading by taking advantage of the independent information in the signal. Classical (spatial) diversity requires that signals are independently fading, hence are (spatially) uncorrelated with each other. In the real world, the received signals are neither totally coherent nor totally uncorrelated. The available diversity is complex and not well understood. In this paper, we study the spatial processing gain (SPG) as a function of the number of receivers used, receiver separation, and array aperture based on experimental data, using beamforming and multichannel combining algorithms. We find that the output symbol signal-to-noise ratio (SNR) for a multichannel equalizer is predominantly determined by the array aperture divided by the signal coherence length, with a negligible dependence on the number of receivers used. For a given number of receivers, an optimal output symbol SNR (OSNR) is achieved by spacing the receivers equal to or greater than the signal coherence length. We model the SPG in decibels as the sum of the noise suppression gain (NSG, equivalent to signal-to-noise enhancement) and the ISI suppression gain (ISG, equivalent to signal-to-ISI enhancement) both expressed in decibels; the latter exploits the spatial diversity and forms the basis for the diversity gain. Data are interpreted using the modeled result as a guide. We discuss a beam-domain processor for sonar arrays, which yields an improved performance at low-input SNR compared to the element-domain processor because of the SNR enhancement from beamforming many sensors.     

Using ocean ambient noise for array self-localization and self-synchronization

Estimates of the travel times between the elements of a bottom hydrophone array can be extracted from the time-averaged ambient noise cross-correlation function (NCF). This is confirmed using 11-min-long data blocks of ambient noise recordings that were collected in May 1995 near the southern California coast at an average depth of 21 m in the 150-700 Hz frequency range. Coherent horizontal wavefronts emerging from the time derivative of the NCF are obtained across the array's aperture and are related to the direct arrival time of the time-domain Green's function (TDGF). These coherent wavefronts are used for array element self-localization (AESL) and array element self-synchronization (AESS). The estimated array element locations are used to beamform on a towed source.     

A study on propeller noise source localization in a cavitation tunnel

Localizing noise sources in cavitation experiments is an important research subject along with predicting noise levels. A cavitation tunnel propeller noise localization method is presented. Propeller noise measurement experiments were performed in the MOERI cavitation tunnel. To create cavitating conditions, a wake-generating dummy body was devised. In addition, 10 hydrophones were put inside a wing-shaped casing to minimize the unexpected flow inducing noise around the hydrophones. After measuring both of the noises of the rotating propeller behind the dummy body and acoustic signals transmitted by a virtual source, the data were processed via three objective functions based on the ideas of matched field processing and source strength estimation to localize noises on the propeller plane. In this paper, the measured noise analysis and the localization results are presented. Through the experiments and the analysis, it was found that the source localization methods that have been used in shallow water applications could be successfully adapted to the cavitation tunnel experiments.     

Multipath channel identification with wavelet packets

This paper deals with the basic modeling problem in underwater acoustics that is the characterization of the channel between a transmitter and a receiver. The problem is analyzed here using an array of sensors that receive PSK signals emitted by several sources. Data come from an experiment realized by a physical system situated in the Mediterranean Sea. In order to identify the multipath channel, we need to access the propagation time delay and the angle of arrival of each propagation ray. However, many of these acoustic ray paths are too close to be separated by classic processing methods (matched filter, beamforming, etc.); new methods with better resolution must be applied in order to analyze the experimental signals and to determine their arrival time on the array of sensors. After a presentation of this problem, we will first discuss high-resolution methods that are usually applied in the localization problem; we will then focus on wavelet packet analysis which provides good results by improving the temporal resolution of acoustic signals     

Ambient Noise Analysis of Deep-Ocean Measurements in the Northeast Pacific

During the late 1960s and throughout the 1970s, the U.S. Navy conducted a series of ocean acoustic measurement exercises to support development of systems and techniques to detect nuclear submarines. The exercises and most of the technical documentation were classified. In 2003, a project was sponsored by the U.S. Office of Naval Research (ONR, Arlington, VA) to declassify documentation and demonstrate the capability to recover acoustic data recorded on magnetic tape. One of the exercises, known as CHURCH OPAL, was selected for demonstration of acoustic data recovery. The record on magnetic tape spanned a period of ten days in September 1975 from a vertical assembly of hydrophones at a site midway between Hawaii and California. This paper presents selected excerpts from a key report (Wittenborn, 1976) on ambient noise that previously was unpublished and unavailable for general distribution. The earlier work is augmented with more complete and detailed analyses of the recovered digital data using modern analytical techniques. Data acquired from the hydrophones below critical depth enabled isolation of ambient noise due to distant shipping and local wind. The frequency band of the acoustic analyses was 30-500 Hz. The wind component of the ambient noise was evaluated at frequencies lower than reported by Wenz (1962).