基于四阶累积量的被动测向声纳高分辨力算法研究

利用高阶累积量的阵列扩展和抑制高斯噪声特性,提出1种基于四阶累积量的逆波束形成算法,并通过滑动平均进一步改善了算法。仿真结果和实验数据表明:逆波束形成算法较常规波束形成方法在空间方位分辨力和噪声抑制能力上明显提高,可以更有效地应用于被动测向声纳中。     

基于二阶锥规划的空域维纳滤波

提出了基于二阶锥规划的空域维纳滤波。基于噪声与阵列流型不相关的特性,将阵列的方向向量作为期望通过二阶锥规划方法对阵列快拍进行维纳滤波。利用维纳滤波抑制噪声的能力,提高阵列的信号检测和方位估计性能。通过仿真研究,验证了单目标时,其低信噪比下的信号检测性能优于Bartlett波束形成,其方位估计的信噪比门限要低于Bartlett波束形成和MUSIC算法,因而是一种优良的波束形成器。该波束形成器用于多波束测深仪和侧扫声纳等海洋测绘设备的阵列,可以有效的提高设备的作用距离和测量精度。     

基于二阶锥规划的空域维纳滤波

提出了基于二阶锥规划的空域维纳滤波。基于噪声与阵列流型不相关的特性,将阵列的方向向量作为期望通过二阶锥规划方法对阵列快拍进行维纳滤波。利用维纳滤波抑制噪声的能力,提高阵列的信号检测和方位估计性能。通过仿真研究,验证了单目标时,其低信噪比下的信号检测性能优于Bartlett波束形成,其方位估计的信噪比门限要低于Bartlett波束形成和MUSIC算法,因而是一种优良的波束形成器。该波束形成器用于多波束测深仪和侧扫声纳等海洋测绘设备的阵列,可以有效的提高设备的作用距离和测量精度。     

一种基于稳健自适应波束形成的声隔离技术

介绍了一种基于稳健自适应波束形成的声隔离技术。通过分析声诱饵工作时干扰与信号的主要入射方位,建立了边发边收声诱饵模型。针对 STMV 波束形成器在基阵模型失配情况下容易出现信号自消的现象,研究了一种解决波束形成稳健性的新方法,即 Robust Capon Beamforming(RCB)算法,实现有效的干扰抑制和信号提取。仿真和试验数据表明：该方法对端射方向干扰具有良好的抑制效果,且能无失真提取出目标信号,为稳健自适应波束形成在声诱饵边发边收中的应用提供了参考。     

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。     

基于高阶累积量的简正波声场匹配场定位效果分析

由于传统情况,匹配场处理器主要针对接收信号为高斯信号,而海洋环境中存在大量的有色噪声,因此设计基于高阶累积量的匹配场处理器具有一定价值。本文介绍了一种在海洋声层析环境下用高阶累积量来降低匹配场处理中高斯有色噪声的算法,对比研究了基于高阶累积量的简正波声场匹配场处理器、传统线性匹配场处理器、自适应匹配场处理器及二阶MUSIC算法匹配场处理器在相同声场环境条件下的匹配场定位效果。研究结果表明,在不考虑参数失配与计算量前提下,接收信号经过基于高阶累积量的简正波声场匹配场处理器后再进行匹配场定位,能有效降低旁瓣,且使用高阶累积量可显著消除海洋传播环境中高斯有色噪声和高斯白噪声对信号的干扰,其匹配效果优于另外三种匹配场处理器。     

圆阵低频超增益波束形成用于水下目标探测研究

针对小尺度基阵难以取得低频空间增益的问题，研究了圆阵超增益波束形成技术检测低频目标的方法。利用小尺度基阵低频噪声的空间相关性，求出基阵的相关系数矩阵，该矩阵对常规波束形成的权系数进行加权，产生超增益波束形成器的最优加权向量。计算机仿真与实测数据处表明：超增益波束形成技术能够在较低的工作频率上形成波束，获得更好的探测定位性能，有良好的应用前景。     

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。     

浅水多波束测深仪高分辨率波束形成方法研究

常规波束形成技术以其稳健性好、计算量低等特点得到广泛应用,但其空间分辨率受阵元个数限制,不能突破瑞利限。因此,常规波束形成技术应用于高精度浅水多波束测深仪时,存在空间分辨率不足和旁瓣干扰等问题。对比研究了最小方差无失真(MVDR)及多重信号分类(MUSIC)波束形成在浅水多波束测深仪中的应用,给出了浅水多波束测深仪的常规波束形成、MVDR和MUSIC处理方法,并结合能量法和相位法两种底检测测深算法,处理了iBeam8120浅水多波束测深仪外场数据,验证了本文方法的性能。     

利用衰减器实现的相控波束形成

文章讨论了相控阵多普勒计程仪波束形成原理和数控衰减器混频及移相原理 ,提出了利用衰减器实现相控波束形成的方法。实验证明了数控衰减器实现相控波束形成器的可行性。     

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     

The TAP III beamforming system

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.     

Spatio-temporal processing of coherent acoustic communication datain shallow water

Achieving reliable underwater communication in shallow water is a difficult task because of the random time-varying nature of multipath propagation. When the product of Doppler-related signal bandwidth spread and multipath-related time spread of the channel is larger than one, some types of adaptive signal processing may not work very well. In this paper, various methods of coherent space-time processing are compared for a condition of a marginally overspread channel operating at 50 kHz. Various combinations of suboptimal spatially adaptive and time adaptive methods are considered. The coherent path beamformer (CPB) and recursive least squares (RLS) adaptive beamformer, both in combination with RLS time filtering, are analyzed. Also considered in the analysis is the combined RLS space-time optimal adaptive processor. Many experiments using broad-band phase-shift-keyed transmissions in shallow water have been conducted to provide data for testing these various processing methods. Because of the rapid time variation of the multipath, the product of bandwidth spread and time spread at this test site approached unity. In this environment, a suboptimal approach consisting of the adaptive beamformer followed by RLS equalization reduced reverberation and transmission errors     

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.     

A reconfigurable real-time interpolation beamformer

This paper describes an architecture for a compact real-time time-delay sonar beamformer. The design is based on a time-domain interpolation concept, and includes shading, interpolation, and time-delay functions for up to 64 sensor data inputs. A one card per beam configuration has been achieved. The digital interpolation technique used improves the effective sampling rate by a factor of 128. The beamformer design features 16-bit arithmetic precision, up to approximately a 5-kHz sampling rate for each of the 64 sensor input channels, and adaptability to any type of array, e.g., linear, circular, or arrays with nonequal interelement spacings. A 31-beam beamformer system, performing more than 620-million arithmetic operations a second, is discussed.     

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 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     

Finite-time performance of a quadratically constrained MVDRbeamformer

The finite-time behavior of a quadratically constrained version of the minimum-variance, distortionless-response line-array beamformer is examined. An entity known as the weight ratio is defined and it is demonstrated that there are simple expressions that relate the weight ratio to the field scenario which explicitly take into account the finite-time misadjustment phenomenon. It is further demonstrated that holding the weight ratio to a value of about 0.5 produces an adaptive beamformer with a beneficial combination of robustness to nonideal situations (e.g., sensor misalignment, pointing errors, correlated interferers, misadjustment effects, etc.) while retaining the ability to adapt quickly in highly dynamic fields     

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.     

Accurate Acoustic Signal Parameter Estimation for Marine Geodesy Surveys

Acoustic signal parameter estimation is important for diverse marine geodesy surveys and several other applications. However, the received signal from a far-field target characterized by planar wavefront propagation is frequently affected by strong nearby interfering signals. Their presence deteriorates the performance of direction-of-arrival (DOA) estimation for far-field target. In order to enhance the reception of signal from far-field target, the near-field/far-field (NFFF) beamformer is proposed. Such a beamformer optimizes beam pattern for far-field detection by maximizing beamformer output in the direction of the far-field target with the imposed condition to eliminate interfering signals generated in near-field locations. As the interference suppression only occurs at the position of near-field interference, a possible blind zone for far-field detection in conventional methods is not created. The NFFF beamformer is applicable for coherent signals and the scenario with multi interferences. For stationary situation where interferers locations are fixed, the NFFF beamformer computations do not require time updates with associated computational load. Furthermore the proposed method can be extended to several new situations such as acoustic monitoring performed from a stationary platform subjected to water currents, waves, winds and other variables, all of them generating nearby interferences and also to different array configurations including 2D and 3D arrays.