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.     

海面噪声场条件下声压与振速的空间互相关函数

介绍了海面噪声场条件下的声压与质点振速的时空相关函数。论证了海面噪声场垂直与水平方向噪声的各向异性程度。结果表明,在垂直于海平面的方向上,噪声场表现为各向异性;在与海面的水平方向上,噪声场表现为各向同性。结果表明,各向同性噪声场条件下基于矢量水听器被动检测的声纳系统目标水平方位角估计是基本可靠的,但垂直方位角的计算需要修正。     

Large aperture digital acoustic array

A digital array of 120 acoustic channels 900 m in length has been constructed to study low-frequency (20-200 Hz) ambient noise in the ocean. The array may be deployed vertically or horizontally from the research platform FLIP and the array elements are localized with a high-frequency acoustic transponder network. The authors describe the instrumentation, telemetry, and navigation systems of the array during a vertical deployment in the northeast Pacific. Preliminary ambient noise spectra are presented for various array depths and local wind speeds. Ambient noise in the frequency band above 100 Hz or below 25 Hz increases with local wind speed. However, in the frequency band 25-100 Hz, ambient noise is independent of wind speed and may be dominated by shipping sources     

A Portable Matched-Field Processing System Using Passive Acoustic Time Synchronization

A portable matched-field processing (MFP) system for tracking marine mammals is presented, constructed by attaching a set of autonomous flash-memory acoustic recorders to a rope to form a four-element vertical array, or "insta-array." The acoustic data are initially time-synchronized by performing a matched-field global inversion using acoustic data from an opportunistic source, and then by exploiting the spatial coherence of the ocean ambient noise background to measure and correct for the relative clock drift between the autonomous recorders. The technique is illustrated by using humpback whale song collected off the eastern Australian coast to synchronize the array, which is then used to track the dive profile of the whale using MFP methods. The ability to deploy autonomous instruments into arbitrary "insta-array" geometries with conventional fishing gear may permit nonintrusive array measurements in regions currently too isolated, expensive, or environmentally hostile for standard acoustic equipment     

Mid- to High-Frequency Acoustic Penetration and Propagation Measurements in a Sandy Sediment

During the recent 2004 Sediment Acoustics Experiment (SAX04), a buried hydrophone array was deployed in a sandy sediment near Fort Walton Beach, FL. The array was used to measure both the acoustic penetration into the sediment and sound speed and attenuation within the sediment while a smaller, diver-deployed array was also used to measure sound speed and attenuation. Both of these systems had been deployed previously during the 1999 Sediment Acoustics Experiment (SAX99). In that experiment, the buried array was used to make measurements in the 11–50-kHz range while the diver-deployed array made measurements in the 80–260-kHz range. For the SAX04 deployment, the frequency range for the measurements using the buried array was lowered to 2 kHz. The diver-deployed array was also modified to cover the 40–260-kHz range. Unlike the SAX99 deployment, there were no obvious sand ripples at the SAX04 buried array site at the time of the measurements. To examine the role of sand ripples in acoustic penetration over this new frequency range, artificial ripple fields were created. For the high frequencies, the penetration was consistent with the model predictions using small-roughness perturbation theory as in SAX99. As the frequency of the incident acoustic field decreased, the evanescent field became the dominant penetration mechanism. The sound speed measured using the buried array exhibits dispersion consistent with the Biot theory while the measured attenuation exceeds the theory predictions at frequencies above 200 kHz.      

Inversion for Time-Evolving Sound-Speed Field in a Shallow Ocean by Ensemble Kalman Filtering

In the context of the recent Maritime Rapid Environmental Assessment/Blue Planet 2007 sea trial (MREA/BP07), this paper presents a range-resolving tomography method based on ensemble Kalman filtering of full-field acoustic measurements, dedicated to the monitoring of environmental parameters in coastal waters. The inverse problem is formulated in a state–space form wherein the time-varying sound-speed field (SSF) is assumed to follow a random walk with known statistics and the acoustic measurements are a nonlinear function of the SSF and the bottom properties. The state–space form enables a straightforward implementation of a nonlinear Kalman filter, leading to a data assimilation problem. Surface measurements augment the measurement vector to constrain the range-dependent structure of the SSF. Realistic scenarios of vertical slice shallow-water tomography experiments are simulated with an oceanic model, based on the MREA/BP07 experiment. Prior geoacoustic inversion on the same location gives the bottom acoustic properties that are input to the propagation model. Simulation results show that the proposed scheme enables the continuous tracking of the range-dependent SSF parameters and their associated uncertainties assimilating new measurements each hour. It is shown that ensemble methods are required to properly manage the nonlinearity of the model. The problem of the sensitivity to the vertical array (VA) configuration is also addressed.      

Effects of sensor placement on acoustic vector-sensor arrayperformance

We consider the role played by the sensor locations in the optimal performance of an array of acoustic vector sensors, First we derive an expression for the Cramer-Rao bound on the azimuth and elevation of a single far-field source for an arbitrary acoustic vector-sensor array in a homogeneous wholespace and show that it has a block diagonal structure, i.e., the source location parameters are uncoupled from the signal and noise strength parameters. We then derive a set of necessary and sufficient geometrical constraints for the two direction parameters, azimuth and elevation, to be uncoupled from each other. Ensuring that these parameters are uncoupled minimizes the bound and means they are the natural or “canonical” location parameters for the model. We argue that it provides a compelling array design criterion. We also consider a bound on the mean-square angular error and its asymptotic normalization, which are useful measures in three-dimensional bearing estimation problems. We derive an expression for this bound and discuss it in terms of the sensors' locations. We then show that our previously derived geometrical conditions are also sufficient to ensure that this bound is independent of azimuth. Finally, we extend those conditions to obtain a set of geometrical constraints that ensure the optimal performance is isotropic     

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.      

Deep-ocean vertical noise directionality

The structure of beam noise measured at the output of a vertical array in a range dependent ocean basin was investigated using the modified wide-angle parabolic equation (PE). Noise sources were distributed throughout the basin, and the field due to each noise source at an array located in the midbasin was calculated. The response of the array to the superposition of the noise sources was found by beamforming. An efficient and direct approach that superimposes the noise sources on the PE field as the field is marched toward the array was developed. Downslope calculations of the midbasin vertical directionality were made between 50 and 400 Hz with this technique. Use of a geoacoustic model shows that the bottom behaves as a low-pass filter     

A circular passive synthetic array: an inverse problem approach

Based on the general concept of the inverse acoustic radiation problem, the temporal scanning of a stationary acoustic field along a closed contour is used to simplify the measurement approach for obtaining information on source directionality. The mathematical formulation is derived from a model of the two-dimensional acoustic field. The formulation of the inverse problem is also investigated to establish a methodology for improving the angular resolution of the array processing. The fundamental relationship between the sound sources and the circular passive synthetic array is explored, utilizing existing mathematical methods, in order to develop the processing algorithm. Other subjects of practical interest, such as directional ambiguity, effect of Doppler frequency, interference noise, and processing gain are discussed. It is concluded that the results can be used to establish guidelines for engineering design and deployment of this type of synthetic array, and to further exploit the new array signal processing technique     

Approximations to directivity for linear, planar, and volumetricapertures and arrays

Even after decades of sonar design, approximations to the directivity factor (DF) or index of an array, are often used inappropriately. Many of the approximations commonly used provide accurate directivity approximations for only the simplest of array geometries. As the array's size, shape, weighting, and complexity increase, there is a renewed need for better directivity approximations. Directivity is defined as the ratio of the output signal-to-noise (SNR) of an array to the input SNR at an omnidirectional element in a spherically isotropic noise field. Calculation of directivity is obtained by integrating the magnitude-squared response of the array over all angles of incidence. In spherical coordinates, these arrival angles are denoted by an azimuthal angle &thetas; and a polar angle φ. Hence, calculation of the directivity requires a two-fold integration over the angular space defined by the azimuthal and polar angles. For complex, large-size arrays consisting of thousands of array elements, directivity calculations using numerical integration procedures can be time consuming, even on state-of-the-art computing systems. This report provides a number of accurate formulas for estimating the directivity of linear, planar, and volumetric apertures and arrays, which are allowed to have arbitrary shading coefficients, steering angles, and directional array element responses     

Acoustic positioning for an array of freely drifting infrasonic sensors

Initial testing of the prototype element of a freely drifting infrasonic sensor array is described. The intent of this measurement system is to gather wide aperture data sets which will be used both to characterize ambient noise in the region 1-10 Hz and to assess the gains possible from beam forming utilizing a collection of very low frequency (VLF) sensors. Coherent processing (beam forming) of the infrasonic sensor data is made possible by relative position measurements derived from mutual acoustic interrogation of the elements at a higher frequency. Surface echo data from a recent sea test of the prototype buoy are used to illustrate the type of pulse processing which will be implemented as a first step in the localization procedure.     

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.     

Acoustic Propagation Through Clustered Bubble Clouds

One of the underlying assumptions in the effective medium theory describing the propagation of acoustic waves through bubble clouds is that the probability of an individual bubble being located at some position in space is independent of the locations of other bubbles. However, bubbles within naturally occurring clouds may be influenced by the dynamics of the fluids in which they are entrained so that they become preferentially concentrated, or clustered, leading to statistical dependence in their positions. For bubble clouds in which the important scattering terms include those with interactions between at least two bubbles, statistical dependence between bubble positions leads to a reduction in the attenuation of the coherent acoustic pressure field from that which would be predicted for a nonclustered bubble cloud. Bubble clustering can be accommodated in effective medium theories using correlation functions describing the relationship between the positions of the bubbles. For double scattering, the two-bubble correlation (i.e., the pair correlation function) must be used, for triple scattering, the three bubble correlation must be used, and so on. In contrast to the three attenuation of the coherent field, making the assumption of independent bubble positions leads to an underestimate of the incoherent field. Both the coherent and incoherent acoustic fields for bubble clouds exhibiting correlated bubble positions are explored in this paper with the use of numerical simulations.     

Noise source level density due to surf. I. Monterey Bay, CA

Ambient noise measurements made in Monterey Bay, CA, in 1981 were reduced by estimations of wave-breaking noise and the residual noise was combined with modeled transmission loss (TL) to estimate the spectral source level of surf-generated noise. A Hamilton geoacoustic model of the coastal environment was derived and used in a finite-element parabolic equation propagation-loss model to obtain TL values. Estimates of both the continuous, or local, and discrete components of wave-breaking noise intensity were subtracted from the total measured noise field to determine the contribution due to surf only. Surf breaking on a uniform 12.5-km linear section of beach near Ft. Ord was found to be the dominant source of surf-generated noise. Estimated noise source level densities for heavy surf at Ft. Ord beach varied from 138 dB ref. 1 μPa Hz^-1/2 m at 1 m from the source at 50 Hz to 107 dB at 1 kHz, with a slope of about -5 dB per octave. Although these results must be considered as preliminary, since they are based on a small number of measurements, they may he useful for prediction of ambient noise in other littoral regions     

声相关计程仪基阵设计方法

针对声相关计程仪的测速特点,研究了一维、二维基阵的设计方法。一维基阵采用基于约束最小冗余的设计方法,可以获得比均匀线阵大得多的阵列孔径,从而提高基阵的利用率,但约束最小冗余线阵(RMRLA)的设计方法计算量巨大,并不适合二维基阵的设计。在重新定义冗余因子,建立理想位置矢量图模型,提出位置矢量重合率等概念的基础上,实现了适用于声相关测速需求的二维基阵的快速设计。仿真结果证明了该设计方法的可行性。     

Research on Hydrodynamic Interference Suppression of Bottom-Mounted Monitoring Platform with Fairing Structure

In the disturbance of unsteady flow field under the sea, the monitoring accuracy and precision of the bottom-mounted acoustic monitoring platform will decrease. In order to reduce the hydrodynamic interference, the platform wrapped with fairing structure and separated from the retrieval unit is described. The suppression effect evaluation based on the correlation theory of sound pressure and particle velocity for spherical wave in infinite homogeneous medium is proposed and the difference value between them is used to evaluate the hydrodynamic restraining performance of the bottom-mounted platform under far field condition. Through the sea test, it is indicated that the platform with sparse layers fairing structure (there are two layers for the fairing, in which the inside layer is 6-layers sparse metal net, and the outside layer is 1-layer polyester cloth, and then it takes sparse layers for short) has no attenuation in the sound pressure response to the sound source signal, but obvious suppression in the velocity response to the hydrodynamic noise. The effective frequency of the fairing structure is decreased below 10 Hz, and the noise magnitude is reduced by 10 dB. With the comparison of different fairing structures, it is concluded that the tighter fairing structure can enhance the performance of sound transmission and flow restraining.     

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     

Reverberation-derived shallow-water bottom scattering strength

Determinations of acoustic scattering strength for sand bottoms have been made at several different shallow-water areas under downward refracting sound propagation conditions in the frequency decade below 1 kHz. The measurements have been made using explosive sources detonated at mid-water depth and bottom-mounted vertical and horizontal hydrophone line arrays as receivers. The ubiquitous presence of multipaths in shallow water prevents a direct-path scattering geometry, and scattering strength must be extracted from the full reverberation field, which complicates the determination of bottom grazing angle dependence of scattering. The major focus of this paper has been the variation of scattering strength with frequency (integrated over participating bottom angles), though estimates of the angular dependence of scattering strength have been made using the vertical receiving array. Typically the integrated scattering strength for sand bottoms reported (and elsewhere) are found to decrease below 1 kHz and in some instances to exhibit a minimum in the several hundred hertz range. Sand bottom scattering strengths below 1 kHz are significantly lower than those predicted by the Mackenzie formula and the limited angular dependence determinations have been found to be consistent with Lambert's law     

Temporal and azimuthal dependence of sound propagation in shallowwater with internal waves

The short time scale (minutes) and azimuthal dependence of sound wave propagation in shallow water regions due to internal waves is examined. Results from the shallow water acoustics in random media (SWARM-95) experiment are presented that reflect these dependencies. Time-dependent internal waves are modeled using the dnoidal solution to the nonlinear internal wave equations, so that the effects of both temporal and spatial variability can be assessed. A full wave parabolic equation model is used to simulate broadband acoustic propagation. It is shown that the short term temporal variability and the azimuthal dependence of the sound field are strongly correlated to the internal wave field