深海声影区稀疏时延估计与声源测距

研究了深海声影区中经一次海底反射的多途声线到达垂直双水听器的时延差与声源位置的关系,提出了一种稀疏时延估计与声源测距方法。首先利用近海面布放的短间距垂直双水听器接收一定频带的声信号,然后计算接收信号的广义互相关函数,并利用频谱搬移和稀疏解卷积技术提取时延差,最后通过时延差匹配,估计水下声源的距离。仿真实验表明,在4300 m深海中,所提方法能够正确提取多途到达时延差,估计声影区内的声源距离。海试结果表明,当垂直接收孔径分别为21 m和30 m时,声源测距误差分别小于13.6%和8.1%。上述结果表明,所提出的时延估计方法可适应带宽较窄的接收信号,多途到达时延估计参数可用于实现声影区中的水下声源测距。      

字典奇异值分解加权压缩感知多径信号参数估计

为提高水声信道多径参数估计的分辨率,提出了一种基于字典奇异值分解的加权压缩感知算法。对于有源声呐,根据发射信号构造字典,对字典进行奇异值分解,利用大特征值对应的特征向量构造信号子空间,然后使用信号子空间对接收信号进行滤波。对滤波结果进行加权压缩感知参数估计,得出最终时延估计结果。仿真实验表明,所提方法能够对水声多径参数进行超分辨估计,适用于任何脉冲信号。湖试处理结果显示,混响背景下该方法也有较好的多径参数估计性能,能够降低接收数据的噪声成分,提高对水声信道的多径时延、个数和幅度的估计精度。      

多频带期望值最大声信号时延估计EI北大核心CSCD

通过分析复杂环境中不同频带声信号时延估计的差异,提出多频带期望值最大时延估计方法。为了使各频带之间无重叠,该方法采用独立分带划分声信号不同频带,然后计算各频带广义互相关函数,并对子带广义互相关函数建立最大似然模型,最后利用期望值最大算法将多维优化转为一维优化的迭代式,获得最优子带广义互相关函数,在此基础上估计声信号的时延信息。数据仿真和实际实验结果表明,多频带期望值最大化时延估计相较常规时延估计有效估计值的百分比提升了10%,并将最优频带互相关函数应用到该定位算法中,在网格间距为0.3 m时,得到的峰值区域汇聚更明显,定位效果更好。     

基于水声信道传播时延排序的分层空时信号检测

基于分层空时编码的多输入多输出技术是一种极具潜力的高速水声通信技术, 但要实现这种潜力需要复杂的空时信号处理方法, 以抵消来自水声信道的多径干扰和异步到达干扰, 以及叠加在接收端的各层信号之间的干扰. 对低复杂度的空时信号处理方案进行了研究, 提出了一种基于子信道传播时延排序的有序连续干扰抵消信号检测算法, 利用子信道间的传播时延差, 实现可使差错概率最小的最佳检测排序; 给出了利用信道估计, 以极低的计算量确定排序的方法, 从而可以大幅降低信号检测的计算复杂度. 采用低复杂度的单载波频域均衡来抵消水声信道中的码间干扰和异步到达干扰. 仿真结果表明, 基于时延排序的信号处理算法可以获得检测性能的改善, 而且性能增益在高数据率时更加显著. 研究结果表明, 采用有效的信号处理方法可使水声信道中造成信号检测干扰的传播时延成为改善系统性能的有利因素.     

浅海波导中水平接收阵被动时反混响抑制方法研究

使用基于射线简正波异地混响理论建立的水平接收阵接收海底混响模型,提出了水平阵被动时反混响抑制方法。以水平接收阵接收回波信号的某一时刻作为开始,分别对其前一时刻和后一时刻的回波信号做时反算子分解获得两个混响子空间,将这两个混响子空间的平均作为该时刻的混响子空间的一个估计,然后利用投影处理来抑制该时刻接收信号中的混响成分。该方法可以在一定程度上克服由于海底粗糙引起的局部时间(距离)段内接收回波信号的不平稳性,使得估计得到的混响子空间更加接近真实混响子空间。因此,利用其进行被动时反混响抑制可以增强回波信混比,提高对目标的检测和方位估计能力。针对典型浅海波导环境开展的仿真验证了本文方法的有效性。      

一种改进的非整数自适应时延估计方法*

针对基于时延估计的机电设备故障声定位中的低信噪比和脉冲噪声情况,用α稳定分布建模噪声,改进了非整数自适应时延估计方法。共变相关法对观测序列进行时延估计粗测,将得到的估计值作为非整数自适应时延估计器的初值;将共变相关法中共变序列作为时延估计器的输入信号,在最小平均p范数准则下迭代得到非整数时延估计值。共变序列保留了原始序列间的时延信息,削弱了不相关的噪声。计算机仿真对比实验验证了改进的方法在脉冲环境和低信噪比条件下有更好的性能。     

利用深海海底声反射区频域干涉结构的声源深度估计方法

提出了一种基于深海海底声反射区声场频域干涉结构特征的水下宽带近海面声源深度估计方法。该方法通过建立深海海底声反射区到达声场结构模型,推导了垂直阵接收信号波束输出幅度谱的近似表达式,利用幅度谱与声源深度和垂直到达角(俯仰角)之间的周期变化关系,将接收信号映射到深度-垂直到达角域中,实现了对宽带声源的深度估计。仿真实验与影响因素分析验证了该原理的正确性,南海实验结果表明:利用阵长为64 m的垂直短阵接收标定深度为50 m和100 m的双弹信号,得到的深度估计结果同实际声源深度吻合较好,估计误差不超过7%,验证了该方法的有效性。     

基于小波变换的广义相关时延估计算法

为了估计两个在空间上分离的传感器所接收信号之间的时间延迟，本文结合小波变换的特点，提出了一个基于小波变换的广义相关时延估计算法并给出了它的时域和频域公式，此算法与直接互相关法相比，能对信号与噪声的假设条件放宽且在低信噪比下能有效估计时延。为了提高时延估计精度，在广义相关时延估计的基础上本文又提出了基于基小波的二次加权法。经仿真证明了这两种算法的有效性。     

随机共振法估计二水听器时延差

针对二元阵航船目标方位估计问题,提出了一种基于随机共振滤波的航船辐射噪声双水听器时延差估计方法。设计了一种随机共振滤波器,提取航船螺旋桨旋转噪声在水听器接收信号中出现的本地时间,并通过滤波结果的互相关来估计目标辐射噪声分别到达两个水听器的时延差。在5 dB的输入信噪比下进行200次仿真,提出方法时延估计的归一化均方误差要低于互谱方法一个数量级。对于海试中的非合作航船,提出的方法可以在1 km距离下准确估计出目标辐射噪声到达双水听器的时延差,从而为在二元阵上实现对航船目标的准确测向提供基础。      

声速剖面对深海远程声脉冲结构的影响

利用东印度洋和南海海域进行的深海远程声传播实验数据,比较分析了声道轴附近深度发射的声信号在两个海域不同声速剖面结构下的远程传播损失和脉冲时间到达结构。通过对比观测发现,两海域的深海声传播损失特性存在一定的差异,声脉冲时间到达结构差异性显著。首先,在东印度洋实验中观测到潜标垂直阵同一接收距离上,靠近声道轴传播的声能量较大,且声道轴附近声速较小但沿其传播的声信号却最先到达,而偏离声道轴传播的声信号延后到达,在整个接收深度上呈现出声道轴附近接收波形早于其他深度到达的分支结构,这与南海典型深海环境下的脉冲时间到达结构存在显著差异。其次,结合深海声道的参数化数学模型,分析了声速剖面对远程脉冲传播时间到达结构的影响机理,并理论解释了两个海域实验中观测到的脉冲声信号时间到达结构现象,其形成原因在于深海声道中决定声速剖面结构的声道轴系列参数的差异。该研究结果对通信声呐在不同海域深海远程环境下的应用具有一定的参考意义。     

基于L型延迟阵列调制宽带转换器的信号载频和二维到达角联合估计

针对现有的基于欠采样的频率和二维到达角的联合估计存在结构复杂问题,本文提出了一种基于调制宽带转换器技术的L型延迟阵列接收结构.利用延迟通道与未延迟通道采样值之间的相位差可直接估计载频,进而计算二维到达角,无需额外的参数配对操作,避免了配对步骤引入的误差和复杂度的提升.并结合所提L型延迟阵列结构的特点构造相关矩阵和三线性模型,提出了两种参数估计算法,一种基于旋转不变子空间算法,计算量小,适用于需要实时处理的场景;另一种基于正则分解技术,鲁棒性较好,适用于信噪比较低的应用场景.仿真实验表明该方法能较好地从欠奈奎斯特样本中估计目标的载频和二维到达角参数.     

Restoration of spectral characteristics of ionospheric wave disturbances

The problem of short-wave radio-signal reflection from a parabolic ionospheric layer with traveling ionospheric disturbances (TID) in the form of density waves is solved in an approximation of geometrical optics. Formulas that relate the arrival angle and Doppler shift of a radio signal to the TID parameters are derived by the small-perturbation method for the characteristic function. The inverse problem of determination of the TID spectral characteristics is solved. The amplitude, dispersion law, and propagation direction of ionospheric irregularitiesare restored.     

Active stereo sound localization

Estimating the direction of arrival of sound in three-dimensional space is typically performed by generalized time-delay processing on a set of signals from a fixed array of omnidirectional microphones. This requires specialized multichannel A/D hardware, and careful arrangement of the microphones into an array. This work is motivated by the desire to instead only use standard two-channel audio A/D hardware and portable equipment. To estimate direction of arrival of persistent sound, the position of the microphones is made variable by mounting them on one or more computer-controlled pan-and-tilt units. In this paper, we describe the signal processing and control algorithm of a device with two omnidirectional microphones on a fixed baseline and two rotational degrees of freedom. Experimental results with real data are reported with both impulsive and speech sounds in an untreated, normally reverberant indoor environment.     

Modal parameter identification through Gabor expansion of response signals

A new method of identifying modal parameters by decomposing response signals with Gabor transform is presented in this paper to estimate natural frequencies, damping ratios and mode shapes of linear time invariant systems. According to Gabor expansion theory, responses of a multi-degree-of-freedom system can be decomposed into uncoupled signal components, each vibrating at a single natural frequency. From these uncoupled signals, modal parameters are subsequently extracted with common methods. The proposed method can process stationary and non-stationary responses and requires no input signal except for the response signals generated by unknown excitation acting on a system. In the sense of less restriction on the in-out signals, the approach based on time-frequency decomposition is very general. A simulation study on a simply supported beam under non-stationary excitation has demonstrated that the proposed method is effective in parameter estimation.     

Optimal Passive Source Localization for Acoustic Emissions

Acoustic emission is a non-destructive testing method where sensors monitor an area of a structure to detect and localize passive sources of elastic waves such as expanding cracks. Passive source localization methods based on times of arrival (TOAs) use TOAs estimated from the noisy signals received by the sensors to estimate the source position. In this work, we derive the probability distribution of TOAs assuming they were obtained by the fixed threshold technique—a popular low-complexity TOA estimation technique—and show that, if the sampling rate is high enough, TOAs can be approximated by a random variable distributed according to a mixture of Gaussian distributions, which reduces to a Gaussian in the low noise regime. The optimal source position estimator is derived assuming the parameters of the mixture are known, in which case its MSE matches the Cramér–Rao lower bound, and an algorithm to estimate the mixture parameters from noisy signals is presented. We also show that the fixed threshold technique produces biased time differences of arrival (TDOAs) and propose a modification of this method to remove the bias. The proposed source position estimator is validated in simulation using biased and unbiased TDOAs, performing better than other TOA-based passive source localization methods in most scenarios.     

混沌噪声背景下微弱脉冲信号的检测及恢复

构建了一种在混沌噪声背景下检测并恢复微弱脉冲信号的模型.首先,基于混沌信号的短期可预测性及其对微小扰动的敏感性,对观测信号进行相空间重构、建立局域线性自回归模型进行单步预测,得到预测误差,并利用假设检验方法从预测误差中检测观测信号中是否含有微弱脉冲信号.然后,对微弱脉冲信号建立单点跳跃模型,并融合局域线性自回归模型,构成双局域线性(DLL)模型,以极小化DLL模型的均方预测误差为目标进行优化,采用向后拟合算法估计模型的参数,并最终恢复出混沌噪声背景下的微弱脉冲信号.仿真实验结果表明本文所建的模型能够有效地检测并恢复出混沌噪声背景中的微弱脉冲信号.     

深海脉冲信号簇到达结构特征及其在水下声源定位中的应用_

为了实现对深海水下声源的定位,对典型深海环境中的脉冲信号到达结构特征进行了理论分析,给出了声源和接收位置位于近海面时到达信号的簇信号形式近似表达式.当声源和接收位置处于近海面深度时,接收到的信号呈簇状结构形式.在提取到达信号中各簇信号到达时间、幅值等特征参数的基础上,提出了一种通过对到达信号中簇信号特征参数匹配搜索进行...     

Estimation of ultrasound attenuation and dispersion using short time Fourier transform

Determination of the acoustic attenuation and dispersion has important applications in ultrasound tissue characterization and non-destructive material testing. Current signal processing methods Fourier transform of ultrasound signals to get the spectra of amplitude and phase to estimate respectively the attenuation and dispersion of a given medium. These methods are frequency domain method and obsessed with ambiguity issue in the phase unwrapping calculation. Conventional ultrasound velocity measuring method detects the time of arrival of a pulse (or echo) signal, which is a time domain method to compute group velocity (not phase velocity). This paper presents a novel approach based on the short time Fourier transform (STFT)--a time-frequency analysis, to estimate the ultrasonic dispersion and attenuation. Only the amplitude information of the pulse-signal spectra is used. Based on the time-frequency presentation, the attenuation coefficient of the signal is obtained by computing the amplitude decay of pulse spectrum in time domain, while phase velocities are obtained based on the "time-of-flight" (TOF) of the mono frequency component of the pulse signals. As a result, we eliminate the ambiguity issue in phase angle calculation. Furthermore, the proposed method makes the phase velocity pedagogically intuitive for novice users. The paper presents experiments to evaluate demonstrate the performance of the proposed method.     

利用多角度海底反向散射信号进行地声参数估计

针对现有海底地声参数估计方法的不足,提出了利用相控参量阵浅地层剖面仪接收的多角度海底反向散射信号进行地声参数估计的方法,首先利用正下方和斜入射方向上沉积层上、下表面的差频反向散射信号进行沉积层厚度和声速估计,然后利用正下方沉积层上、下表面两个不同频率的差频信号的反向散射信号估计沉积层衰减系数,最后利用正下方沉积层上表面原频反向散射信号估计沉积层阻抗,计算沉积层密度从而解决和声速的耦合性,通过水池试验验证了该方法的有效性。     

A reliable acoustic path： Physical properties and a source localization method

<正>The physical properties of a reliable acoustic path(RAP) are analysed and subsequently a weighted-subspacefitting matched field(WSF-MF) method for passive localization is presented by exploiting the properties of the RAP environment.The RAP is an important acoustic duct in the deep ocean,which occurs when the receiver is placed near the bottom where the sound velocity exceeds the maximum sound velocity in the vicinity of the surface.It is found that in the RAP environment the transmission loss is rather low and no blind zone of surveillance exists in a medium range. The ray theory is used to explain these phenomena.Furthermore,the analysis of the arrival structures shows that the source localization method based on arrival angle is feasible in this environment.However,the conventional methods suffer from the complicated and inaccurate estimation of the arrival angle.In this paper,a straightforward WSF-MF method is derived to exploit the information about the arrival angles indirectly.The method is to minimize the distance between the signal subspace and the spanned space by the array manifold in a finite range-depth space rather than the arrival-angle space.Simulations are performed to demonstrate the features of the method,and the results are explained by the arrival structures in the RAP environment.