Motion compensation for adaptive horizontal line array processing

Large aperture horizontal line arrays have small resolution cells and can be used to separate a target signal from an interference signal by array beamforming. High-resolution adaptive array processing can be used to place a null at the interference signal so that the array gain can be much higher than that of conventional beamforming. But these nice features are significantly degraded by the source motion, which reduces the time period under which the environment can be considered stationary from the array processing point of view. For adaptive array processing, a large number of data samples are generally required to minimize the variance of the cross-spectral density, or the covariance matrix, between the array elements. For a moving source and interference, the penalty of integrating over a large number of samples is the spread of signal and interference energy to more than one or two eigenvalues. The signal and interference are no longer clearly identified by the eigenvectors and, consequently, the ability to suppress the interference suffers. We show in this paper that the effect of source motion can be compensated for the (signal) beam covariance matrix, thus allowing integration over a large number of data samples without loss in the signal beam power. We employ an equivalent of a rotating coordinate frame to track the signal bearing change and use the waveguide invariant theory to compensate the signal range change by frequency shifting.     

Time reversal multiple-input/multiple-output acoustic communication enhanced by parallel interference cancellation

Multiple-input/multiple-output (MIMO) techniques can lead to significant improvements of underwater acoustic communication capabilities. In this paper, receivers based on time reversal processing are developed for high frequency underwater MIMO channels. Time reversal followed by a single channel decision feedback equalizer, aided by frequent channel updates, is used to compensate for the time-varying inter-symbol interference. A parallel interference cancellation method is incorporated to suppress the co-channel interference in the MIMO system. The receiver performance is demonstrated by a 2008 shallow water experiment in Kauai, Hawaii. In the experiment, high frequency MIMO signals centered at 16 kHz were transmitted every hour during a 35 h period from an 8-element source array to a wide aperture 16-element vertical receiving array at 4 km range. The interference cancellation method is shown to generate significant performance enhancement, on average 2-4 dB in the output signal-to-noise ratio per data stream, throughout the 35 h MIMO transmissions. Further, communication performance and achieved data rates exhibit significant changes over the 35 h period as a result of stratification of the water column.     

波束形成后多干扰抵消方法

为解决常规波束形成后干扰抵消方法仅能从目标波束内消除一个干扰影响的局限性,提出了一种可实现多干扰抑制的波束形成后干扰抵消方法,用于解决多干扰条件下的弱目标波束形成问题.该方法通过最小化多个干扰波束对目标波束的影响,获得用于抵消干扰波束的加权值;然后从目标波束中减去加权后的干扰波束,从而获得抑制干扰后的目标波束。该方法继承了常规波束形成方法的稳健性,因此在阵列流型失配等存在误差的条件下可以用于目标波束形成。仿真和海上实验数据表明,与最小方差无失真响应方法和解卷积的常规波束形成方法相比,该方法可以有效抑制干扰的影响,且当目标与任一干扰方位相近时,其方位谱中可以观察到目标波束,而另外两种方法的目标波束均被干扰波束或其旁瓣所掩盖;与设置凹槽的波束图综合方法相比,该方法算法复杂度低,计算速度快,稳健性更好。      

Detection of Gaussian signals in Poisson-modulated interference

Passive broadband detection of target signals by an array of hydrophones in the presence of multiple discrete interferers is analyzed under Gaussian statistics and low signal-to-noise ratio conditions. A nonhomogeneous Poisson-modulated interference process is used to model the ensemble of possible arrival directions of the discrete interferers. Closed-form expressions are derived for the recognition differential of the passive-sonar equation in the presence of Poisson-modulated interference. The interference-compensated recognition differential differs from the classical recognition differential by an additive positive term that depend on the interference-to-noise ratio, the directionality of the Poisson-modulated interference, and the array beam pattern.     

极紫外光源锡液滴靶的检测研究

采用正交放置的两路CCD,基于图像采集及处理的方法,建立了极紫外光源锡液滴靶发生装置的锡液滴检测系统,可以实时监测锡液滴的运动状态及稳定性。对本实验室频率为34kHz的锡靶发生器产生的液滴进行了锡液滴监测实验。监测结果显示,锡液滴直径约为137μm,平均间距为375μm,稳定性较好,并对水平面上横向稳定性进行了分析。     

The performance of matched-field track-before-detect methods using shallow-water Pacific data

Matched-field track-before-detect processing, which extends the concept of matched-field processing to include modeling of the source dynamics, has recently emerged as a promising approach for maintaining the track of a moving source. In this paper, optimal Bayesian and minimum variance beamforming track-before-detect algorithms which incorporate a priori knowledge of the source dynamics in addition to the underlying uncertainties in the ocean environment are presented. A Markov model is utilized for the source motion as a means of capturing the stochastic nature of the source dynamics without assuming uniform motion. In addition, the relationship between optimal Bayesian track-before-detect processing and minimum variance track-before-detect beamforming is examined, revealing how an optimal tracking philosophy may be used to guide the modification of existing beamforming techniques to incorporate track-before-detect capabilities. Further, the benefits of implementing an optimal approach over conventional methods are illustrated through application of these methods to shallow-water Pacific data collected as part of the SWellEX-1 experiment. The results show that incorporating Markovian dynamics for the source motion provides marked improvement in the ability to maintain target track without the use of a uniform velocity hypothesis.     

利用宽带声场频率-掠射角干涉结构的深海直达声区目标深度估计方法

针对典型深海环境中宽带声源的深度分辨问题,通过研究深海声场随频率起伏的干涉结构与垂直线阵频域波束输出图中的干涉结构,给出一种直达声区内可区分多水下目标的宽带声源深度估计方法。该方法以近水面目标的射线声场模型为基础,推导出近海面宽带声源接收声场的波束输出表达式,阐明了频域波束输出图中干涉结构与声源深度的对应关系。然后利用改进的傅里叶变换方法将二维频域波束输出图映射到声源深度-掠射角度域,可实现声源深度信息的有效分离。最后开展了深海实验验证,利用垂直阵接收拖曳声源发射的宽带白噪声信号,拖曳声源深度计算结果与实测声源深度基本一致。数值仿真与实验结果均表明该方法可以在多目标复杂环境下准确估计出水下宽带声源的深度。      

Measurement and modeling of the acoustic field near an underwater vehicle and implications for acoustic source localization

The performance of traditional techniques of passive localization in ocean acoustics such as time-of-arrival (phase differences) and amplitude ratios measured by multiple receivers may be degraded when the receivers are placed on an underwater vehicle due to effects of scattering. However, knowledge of the interference pattern caused by scattering provides a potential enhancement to traditional source localization techniques. Results based on a study using data from a multi-element receiving array mounted on the inner shroud of an autonomous underwater vehicle show that scattering causes the localization ambiguities (side lobes) to decrease in overall level and to move closer to the true source location, thereby improving localization performance, for signals in the frequency band 2-8 kHz. These measurements are compared with numerical modeling results from a two-dimensional time domain finite difference scheme for scattering from two fluid-loaded cylindrical shells. Measured and numerically modeled results are presented for multiple source aspect angles and frequencies. Matched field processing techniques quantify the source localization capabilities for both measurements and numerical modeling output.     

Spectrogram and localization of a sound source in shallow water

The paper describes an approach to detecting a sound source and estimating the radial velocity and distance from the receiver, based on repeat Fourier transformation of the interference pattern formed during motion. The obtained spectrogram contains localized domains of the spectral density of single modes. We estimate the localization domain and spectral density distribution and discuss the resolution of moving sound sources. We present the results of a field experiment and consider the interference immunity of the approach for localizing a source using a single receiver.     

Gravitomagnetic time-varying effects on the motion of a test particle

We study the effects of a time-varying gravitomagnetic field on the motion of test particles. Starting from recent results, we consider the gravitomagnetic field of a source whose spin angular momentum has a linearly time-varying magnitude. The acceleration due to such a time-varying gravitomagnetic field is considered as a perturbation of the Newtonian motion, and we explicitly evaluate the effects of this perturbation on the Keplerian elements of a closed orbit. The theoretical predictions are compared with actual astronomical and astrophysical scenarios, both in the solar system and in binary pulsars systems, in order to evaluate the impact of these effects on real systems.     

Improvement in matched field processing using the CLEAN algorithm

Adaptive matched field processing such as the minimum variance distortionless processor (MV) provides excellent sidelobe (or ambiguity) suppression capability in source localization given a perfect knowledge of the ocean environment. Unfortunately, this processing is very sensitive to sources of mismatch and robust adaptive algorithms are then employed such as a white noise constraint (WNC) often at the expense of insufficient sidelobe control. The CLEAN algorithm was introduced in radio astronomy [Astron. Astrophys. Suppl. Ser. 15, 417-426 (1974)] to produce a high quality image of the sky by reducing sidelobe-induced artifacts. In this paper, the CLEAN concept is extended to matched field processing. Numerical simulations and experimental data demonstrate that matched field processing combined with the CLEAN algorithm can improve performance, especially when a weak source is masked by sidelobes from a much stronger source.     

Adaptive autoregressive modeling of non-stationary vibration signals under distinct gear states. Part 1: modeling

Non-parametric time-frequency techniques are increasingly developed and employed to process non-stationary vibration signals of rotating machinery in a great deal of condition monitoring literature. However, their capacity to reveal power variations in the time-frequency space as precisely as possible becomes a hard constraint when the aim is that of monitoring the occurrence of mechanical faults. Therefore, for an early diagnosis, it is imperative to utilize methods with high temporal resolution, aiming at detecting spectral variations occurring in a very short time. This paper proposes three new adaptive parametric models transformed from time-varying vector-autoregressive model with their parameters estimated by means of noise-adaptive Kalman filter, extended Kalman filter and modified extended Kalman filter, respectively, on the basis of different assumptions. The performance analysis of the proposed adaptive parametric models is demonstrated using numerically generated non-stationary test signals. The results suggest that the proposed models possess appealing advantages in processing non-stationary signals and thus are able to provide reliable time-frequency domain information for condition monitoring.     

激光干涉任意转角测量信号的获得及误差补偿技术

采用信号处理的方法对激光干涉测角系统的特性进行线性化处理,需要产生两路正交信号。采用空间互相垂直的两套干涉系统获得信号难免会使系统复杂、调整困难,因此提出了一种插值处理方法,可以利用测量得到的一路信号获得线性输出信号。输出信号的非线性误差较处理前大大减小了,但是无法满足高精度测量的要求。还提出了针对误差产生的原因进行的误差补偿技术。实验结果表明,提出的方法可以实现任意转角的高精度测量,光程差的测量误差小于±0 2μm,对应的转角测量误差为0 63″。     

Experimental studies of applications of time-reversal acoustics to noncoherent underwater communications

The most difficult problem in shallow underwater acoustic communications is considered to be the time-varying multipath propagation because it impacts negatively on data rates. At high data rates the intersymbol interference requires adaptive algorithms on the receiver side that lead to computationally intensive and complex signal processing. A novel technique called time-reversal acoustics (TRA) can environmentally adapt the acoustic propagation effects of a complex medium in order to focus energy at a particular target range and depth. Using TRA, the multipath structure is reduced because all the propagation paths add coherently at the intended target location. This property of time-reversal acoustics suggests a potential application in the field of noncoherent acoustic communications. This work presents results of a tank scale experiment using an algorithm for rapid transmission of binary data in a complex underwater environment with the TRA approach. A simple 15-symbol code provides an example of the simplicity and feasibility of the approach. Covert coding due to the inherent scrambling induced by the environment at points other than the intended receiver is also investigated. The experiments described suggest a high potential in data rate for the time-reversal approach in underwater acoustic communications while keeping the computational complexity low.     

Robust adaptive matched field processing with environmental uncertainty

The main challenges on detection and localization of quiet targets in littoral regions for passive sonar are the complicated acoustic propagation and the prevalence of loud ship interferences on the surface. Adaptive matched field processing can provide the ability to null surface interferences, but the mismatch between the computed and actual array steering vectors due to environment uncertainty, and the motion of both targets and interferences can result in loss of array gain significantly. To address the problem of environmental mismatch and target motion, a robust motion compensation algorithm and a system scheme for adaptive matched field processing have been developed. Both Numerical simulation and analysis of experimental data demonstrates that the robust AMFP scheme could suppress surface loud interferences and improve the detection performance for underwater weak moving targets in complex shallow water.     

Asynchronous multiuser underwater acoustic communications (L)

An asynchronous multiuser system is proposed to support multiple-access underwater communications without the use of code-division multiple-access or a feedback channel. The rich multipath channels experienced by spatially separated users will be sufficient to ensure separation of collided packets at the base station. The iterative receiver will employ a combination of adaptive time-reversal processing, matching pursuit, and successive interference cancellation in a block-wise fashion to achieve multiuser separability. Data collected during the KAM11 experiment are used to illustrate the system's capability in a dynamic, time-varying environment.     

基于时空滤波理论的低频声场干涉结构研究

基于时空滤波理论, 采用相干函数、功率响应函数、扩展分布函数、双频函数四个系统函数较波导不变量更全面地描述了低频声场的相干结构. 理论分析、仿真研究和海试数据处理都验证了低频声场存在稳定的干涉结构, 当目标由远及近、又由近及远做匀速直线运动时, 相干函数(LOFAR图)的干涉结构为一簇类双曲线; 功率响应函数的干涉图表征了简正波群时延的差分, 体现了其频散特性; 扩展分布函数能反映LOFAR图上干涉条纹的斜率大小; 而双频函数能体现相慢度的差分. 各个系统函数均能特别突出干涉结构的某方面的特征, 各有特色.     

Detection of weak signals using fast projection algorithms

The problem of detecting weak signals in complex noise situations using projection adaptive algorithms is considered. The existing algorithms are analyzed, and a novel algorithm oriented at detecting weak signals is proposed. Results of the algorithm’s operation are demonstrated in a simulated noise situation consisting of interference signals with different intensities under the assumption of their multipath propagation and scattering. The proposed algorithm is compared with the well-known classical Capon algorithm, and a significant reduction in the contact loss time as applied to a low-noise target near strong interference sources is demonstrated.     

用于浅海有源声呐目标深度估计的匹配相位处理EI北大核心CSCD

针对浅海波导中有源声呐目标深度估计问题,提出了通过单个声源发射低频宽带信号,垂直双接收水听器接收目标回波,利用宽带目标回波比值的相位特征进行匹配相位处理的方法,不仅消除了目标散射特性对有源声呐目标深度估计的影响,而且仅需计算单程的信道传递函数,运算量小,有利于实时处理的实现需求。首先通过理论推导,定义了一个隐含目标深度、且与目标散射特性无关的声场特征——单程传播向量宽带干涉结构,在Pekeris波导条件下仿真分析了简正波阶数与单程传播向量宽带干涉结构的幅度和相位的关系,发现选取较多的简正波阶数贡献的单程传播向量宽带干涉结构进行匹配相位处理可以提高目标深度估计的性能。进一步仿真分析表明,在发射信号选取一定的时间长度和带宽的条件下,利用全部阶简正波且信噪比大于-10 dB时,方法的深度估计误差在5 m之内。最后分析了信号时间长度和处理带宽对有源声呐目标深度估计性能的影响,以及海底声速、海深和声速剖面失配时方法的鲁棒性。     

A new efficient two-channel backward algorithm for speech intelligibility enhancement: A subband approach

This paper addresses the problem of speech intelligibility enhancement by adaptive filtering algorithms employed with subband techniques. The two structures named the forward and backward blind source separation structures are extensively used in the speech enhancement and source separation areas, and largely studied in the literature with convolutive and non-convolutive mixtures. These two structures use two-microphones to generate the convolutive/non-convolutive mixing signal, and provide at the outputs the target and the jammer signal components. In this paper, we focus our interest on the backward structure employed to enhance the speech signal from a convolutive mixture. Furthermore, we propose a subband implementation of this structure to improve its behavior with speech signal. The new proposed subband-Backward BSS (SBBSS) structure allows a very important improvement of the convergence speed of the adaptive filtering algorithms when the subband-number is selected high. In order to improve the robustness of the proposed subband structure, we have adapted then applied a new criterion that combines the System Mismatch and the Mean-Errors criterion minimization. The proposed subband backward structure, when it is combined with this new criterion minimization, allows to enhance the output speech signal by reducing the distortion and the noise components. The performance of the proposed subband backward structure is validated through several objective criteria which are given and described in this paper.