运动平台混响的时空耦合统计模型

混响时空耦合特性是空时处理混响抑制方法中的关键问题之一。传统方法不是建立在统计模型基础上的,对实际海洋混响扩展、海洋混响环境影响及基阵非理想性导致的混响时空耦合分析缺乏有效性。论文在混响概率模型基础上,推导了运动平台的混响时空耦合关系,进一步给出了复杂水声混响环境及具体阵列指向特性下的混响时空耦合规律的统计模型。利用该模型,推导了理想窄波束基阵混响的时空耦合关系。理论解析结果和计算机仿真清晰地展示了理想阵列流形的混响时空分布规律,验证了本文理论推导结果的正确性。最后采用数值计算方法分析非理想基阵的混响空时耦合关系及空时分布规律,结果表明利用本文的混响空时耦合模型,能获得实际基阵的混响空时分布规律,可为空时处理混响抑制算法与系统设计提供一定的理论支撑。      

水声信号稀疏重构高分辨角度-多普勒成像

针对傅氏空时二维谱估计分辨率低以及声呐空时采样数据样本数不足给角度-多普勒成像带来困难的问题,提出一种水声信号稀疏重构的高分辨角度-多普勒成像方法和抗混响空时滤波器的稀疏重构方法。该方法在声呐阵列单测量向量的极少观测样本条件下,建立阵列信号的空时稀疏表示模型,应用稀疏表示的匹配追踪算法和基追踪算法重构回波与混响的高分辨角度-多普勒像。并根据运动声呐回波与混响的空时分布规律及声呐待检测距离单元位置的先验信息,沿着混响空时分布脊线设计混响稀疏表示的专用空时导向向量字典,通过重构抗混响空时滤波器来抑制角度-多普勒平面的混响干扰。对运动声呐前视和侧视阵列的计算机仿真结果表明,在混响背景中,该方法采用声呐阵列单测量向量重构了低速运动目标多亮点回波的高分辨角度-多普勒像,频率分辨率突破傅里叶分辨率,角度分辨率突破阵列瑞利限,分辨率明显优于傅氏空时谱估计。      

直接数据域局域联合混响抑制方法研究

针对非平稳条件下混响抑制问题,提出一种可利用待测数据统计信息的直接数据域局域联合(D3JDL)空时自适应方法。不同于级联方法,该方法没有独立运行直接数据域空时自适应和局域联合方法,仅在待测空时样本内用直接数据域空时自适应原理构造非期望数据矩阵,用于获取待测样本的统计信息,并作为局域联合方法的输入进行降维空时处理。由于D3JDL方法不存在待测样本与学习样本之间统计特性失配的问题,且较好地利用了统计信息,因此具备良好的非平稳环境适应能力。仿真和实验数据处理结果表明,D3JDL方法抗混响效果优于常规波束形成加匹配滤波处理和其它空时自适应处理方法。      

利用正弦调频信号的宽带速度敏感特性抑制混响

有源声呐探测水中低速目标时,混响与目标回波的频率接近,目标回波与混响分离困难.根据运动目标与产生混响的散射体在速度域的差异,利用正弦调频信号(SFM)的速度敏感特性实现目标回波与混响的有效分离.推导了 SFM信号宽带混响抑制能力与速度分辨模糊之间的约束关系,建立了波形优化设计的搜索模型.通过多普勒滤波处理将数据从频率域...     

Array gain of fourth-order cumulants beamforming under typical probability density background

The fourth-order cumulant of zero mean Gaussian distribution noise always equals to zero theoretically.In practice the probability density of noise and reverberation is the key problem to performance of the fourth-order cumulant beamforming technique.In this paper,the array gain functions of the fourth-order cumulant beamforming are deducted considering the instantaneous amplitude distribution of the ambient sea noise and bottom reverberation respectively.And the relationships are determined between array gain and the factors including the number of the array elements,the fourth-order and second-order statistical properties of the noise and reverberation,and the input signal-to-noise ratio.It is also verified that there is a critical signal-to-interference ratio and the fourth-order cumulant beamforming can obtain higher gain and resolution than the conventional beamforming method when the ratio is larger than it.The results of experiment data processing demonstrate that the gain and the resolution of the fourth-order cumulant beamforming coincide with the theoretic.     

混响室雷达地杂波统计特性模拟

为实现混响室中威布尔分布杂波的有效模拟,营造逼真的雷达电磁环境,基于广义平稳非相关散射电磁环境下混响室信道衰落特性,通过控制幅度调制信号对衰落系数进行补偿,提出一种基于时域波形设计的混响室地杂波模拟方法。通过调节输入信号序列中大、小幅度脉冲比例,改变混响室传统输入信号的零均值幅度特性,以实测地杂波统计特性为参考,最终得到与实测数据统计特性参数一致的混响室杂波电磁环境。用最大似然估计和KS检验法对实验数据作参数估计和假设检验,实测地杂波数据拟合于标准威布尔模型,混响室实验数据拟合于实测地杂波幅度统计模型,从而实现了实测地杂波起伏特性的混响室有效模拟。     

High-resolution angle-Doppler imaging by sparse recovery of underwater acoustic signals

The low resolution of Fourier two-dimensional spatial temporal spectrum estimation and the insufficient sample size of sonar space time sampling data often caused difficulties in high-resolution space time spectrum estimation. Aiming to solve this problem, we proposed a high-resolution angle-Doppler imaging method and designed an anti-reverberation space time filter based on the sparse recovery of underwater acoustic signals. The proposed imaging method established the spatial temporal sparse representation model of array signal under the condition of few observation samples of single measurement vector(SMV), and reconstructed the high-resolution angle-Doppler profile of echo and reverberation through the matching pursuit(MP) algorithm and basis pursuit(BP) algorithm. By utilizing the space time distribution characteristics of echo and reverberation and the prior information of sonar rangecell under test(RUT), a reverberation dictionary composed of special space time guidance vectors was designed, and was used to reconstruct the reverberation and formed an anti-reverberation space time filter to suppress the reverberation interference of angle-Doppler plane. Computer simulations indicated that, under two conditions of forward and side-view array of moving sonar, the single measurement vector of sonar array was successfully used in the reverberation background to reconstruct the high-resolution angle-Doppler profile of low-speed moving target multi-spot echo. Its frequency resolution had a better performance than the Fourier resolution and the angle resolution broke through the Rayleigh limit of the array, the resolution was obviously superior to the Fourier space time spectrum estimation.     

Surface reverberation at sound field focusing in shallow water

A numerical experiment is carried out to study the long-range surface reverberation in the presence of intense surface waves for the case of using vertical transmitting arrays providing sound field focusing at different depths. To focus the field, a phase conjugation of acoustic waves from a probe source positioned at the focusing point is used. It is demonstrated that surface waves considerably affect the focusing quality at a distance of several tens of kilometers from the transmitting array. This prevents the efficient suppression of long-range reverberation by increasing the focusing depth.     

典型概率密度背景下四阶累积量波束形成的阵增益

在理论上零均值高斯分布噪声的四阶累积量恒等于零,而实际噪声和混响的概率密度是影响四阶累积量波束形成技术性能的关键问题。本文针对海洋环境噪声与海底混响的瞬时幅度分布,分别推导了四阶累积量波束形成阵增益函数;建立了阵增益与阵元数、海洋环境噪声和海底混响统计特性的四阶矩和二阶矩及输入信噪比的关系;确定了与常规波束形成阵增益之间存在临界信噪比,高于临界信干比时四阶累积量波束形成可以获得比常规波束形成更高的阵增益与分辨率。实验数据处理结果验证了四阶累积量波束形成阵增益和分辨率理论结果的一致性。      

Kalman filter-based microphone array signal processing using the equivalent source model

This paper demonstrates that microphone array signal processing can be implemented by using adaptive model-based filtering approaches. Nearfield and farfield sound propagation models are formulated into state-space forms in light of the Equivalent Source Method (ESM). In the model, the unknown source amplitudes of the virtual sources are adaptively estimated by using Kalman filters (KFs). The nearfield array aimed at noise source identification is based on a Multiple-Input–Multiple-Output (MIMO) state-space model with minimal realization, whereas the farfield array technique aimed at speech quality enhancement is based on a Single-Input–Multiple-Output (SIMO) state-space model. Performance of the nearfield array is evaluated in terms of relative error of the velocity reconstructed on the actual source surface. Numerical simulations for the nearfield array were conducted with a baffled planar piston source. From the error metric, the proposed KF algorithm proved effective in identifying noise sources. Objective simulations and subjective experiments are undertaken to validate the proposed farfield arrays in comparison with two conventional methods. The results of objective tests indicated that the farfield arrays significantly enhanced the speech quality and word recognition rate. The results of subjective tests post-processed with the analysis of variance (ANOVA) and a post-hoc Fisher's least significant difference (LSD) test have shown great promise in the KF-based microphone array signal processing techniques.     

混响室的概率统计分析方法及其蒙特卡罗模拟

 提出了一种研究混响室问题的理论分析方法,从理想混响室的离散概率模型出发,采用蒙特卡罗模拟来获得室内电磁场量及其相关物理量的概率统计特性。运用该方法获得了混响室内若干场量（均为归一化值）的概率密度函数、室内一段均匀无耗单导体传输线的负载平均耦合截面及负载电流响应模值的概率分布,所得结果与已有文献给出的结果（包括由解析公式计算或测量所得的结果）吻合良好。由于蒙特卡罗方法的普适性,采用该方法可以获得任意待分析参量的全部概率统计特性（只要样本数足够大）。同时该方法无需针对具体混响室进行,只要在其测试区域能较好地与理想混响室的基本概率模型相吻合,就能给出较精确的计算结果,且其计算量远远小于基于全波分析的蒙特卡罗方法。     

Determination of the path of a sound source moving in an inhomogeneous medium

The possibility of determining the actual paths of moving sound sources from the signals received by a linear horizontal array whose size is large compared to the wavelength is investigated with the use of procedures suggested earlier for the imaging of dynamic objects moving below an inhomogeneous layer. Two cases of signal reception are considered: when the signals propagate in the oceanic waveguide and when the signals propagate through an inhomogeneous layer located near the array. It is shown that, unlike the standard spatial processing procedures, the proposed methods allow one to measure the absolute angular displacements to within the diffraction resolution of the array and to eliminate the ambiguity in angular measurements. An important point is that the proposed methods require no prior data on the parameters of the inhomogeneous layer or the multimode waveguide.     

修正的混响多普勒扩展非对称统计模型

运动声源探测低速目标时,混响谱呈现的非对称性导致正负速度目标的回波具有不同的信混比,需要准确估计双边谱的扩展程度来分别设计检测策略,而传统的对称模型不能描述这种非对称性。基于双边指数分布对称模型,修正的非对称统计模型可用来描述由平台运动、海面运动等引起的混响谱非对称扩展现象。文中推导了非对称模型的数学表达式,并阐述了模型各参数的物理意义。运动声呐平台的湖试实验分析结果验证了该模型对平台在不同速度运动时引起的非对称混响谱都有较高的拟合度,证明混响谱负向多普勒扩展程度随平台运动速度提高而加剧,且非对称模型参数具有量化描述混响谱非对称扩展程度的能力。      

Adaptive beamforming for nonstationary arrays

Adaptive beamformers automatically adjust to the input data on the sensors in an attempt to maximize the bearing resolution of a signal and minimize the effects of noise or interfering signals. To the author's knowledge, all adaptive beamformers currently available in the literature assume that the sensors are stationary over the time required to collect the data. This assumption is invalid on arrays of sensors mounted on nonstationary platforms, and results in poor beamforming performance. In this paper we present an adaptive beamformer that has been designed to operate on nonstationary arrays. The beamformer directly incorporates any changes in array shape or heading that may occur during the time required to collect the data. The output of the adaptive beamformer is shown for both synthetic data and for real data collected on a towed array. Results show that signal detection, bearing accuracy, bearing resolution, and interference suppression all increase when the array shape and track are incorporated into the beamformer if the sensor platform is not stationary.     

基于组合线阵的近程运动小目标被动定位跟踪

针对水下运动小目标被动探测与定位问题,提出了分段组合子阵联合处理探测方法,各探测子阵采用相互垂直结构布局,克服了单阵方位模糊及无法定位等问题。研究了分段组合子阵的宽带最小方差无畸变失真响应近场聚焦目标定位方法,实现了近程蛙人等水下小目标被动高精度定位。在此基础上,采用卡尔曼滤波算法对目标运动轨迹进行预测估计,将目标定位信息与跟踪轨迹信息进行匹配,实现高背景噪音下的运动小目标的跟踪处理。理论分析及仿真结果表明,分段子阵联合处理能有效对水下运动小目标进行定位和跟踪,海试试验进一步验证了该方法的有效性。     

Statistical model of clutter suppression in tissue harmonic imaging

A statistical model is developed for the suppression of clutter in tissue harmonic imaging (THI). Tissue heterogeneity is modeled as a random phase screen that is characterized by its correlation length and variance. With the autocorrelation function taken to be Gaussian and for small variance, statistical solutions are derived for the mean intensities at the fundamental and second-harmonic frequencies in the field of a focused sound beam that propagates through the phase screen. The statistical solutions are verified by comparison with ensemble averaging of direct numerical simulations. The model demonstrates that THI reduces the aberration clutter appearing in the focal region regardless of the depth of the aberrating layer, with suppression of the clutter most effective when the layer is close to the source. The model is also applied to the reverberation clutter that is transmitted forward along the axis of the beam. As with aberration clutter, suppression of such reverberation clutter by THI is most pronounced when the tissue heterogeneity is located close to the source.     

圆柱阵子阵分级处理的稳健超指向性波束形成方法

提出了圆柱阵子阵分级处理的稳健超指向性波束形成方法。首先建立了圆柱阵分两级子阵进行波束形成的信号模型,接着利用空间均匀噪声场中噪声互谱矩阵的循环特性得到基于特征波束分解与综合模型的圆柱阵超指向性的最优解,然后仿真研究了其误差敏感度函数、阵增益和波束图等性能指标,并与圆柱阵的传统全局处理方法进行了对比。提出的分两级子阵处理的超指向性方法与传统全局处理方法相比不仅降低了数据存储量和波束形成计算量,而且进一步提升了稳健性,并且在低频段的阵增益远远高于常规波束形成的值,对水下声呐阵列的设计具有一定的参考价值。      

A localization approach for multiple sound sources via an expectation maximization algorithm using differential microphone arrays

Conventional sound localization approaches with small-sized microphone arrays are usually sensitive to noise and reverberation. To deal with the problem, an approach based on expectation maximization algorithm with differential microphone arrays(DMAs) is proposed.Firstly, the parameters of Gaussian mixture model for time-frequency instantaneous direction estimation are estimated through the EM algorithm, and then the direction of each sound source is estimated via time-frequency separation. In order to overcome the weakness of existing time-frequency separation techniques, an improved method, which combines the advantages of both the hard and soft separation methods, is also proposed. The improved time-frequency separation method is shown to be less sensitive to noise and reverberation. Simulation and experimental results demonstrate that the proposed localization approach is superior to its existing counterparts in terms of localization accuracy and robustness.     

Room acoustical model of external reverberation

A theory of external reverberation in urban built-up environments is developed, based on a classical room acoustical model. In the model, external reverberation is analyzed as a special limiting case of internal reverberation in rooms. Explicit formulae are deduced for the statistical value of the external reverberation time, and the spatial distribution of the external sound field amplitude with distance from a fixed, constant power, sound source, for which comparison with published experimental results is possible. Predictions of the theory compare reasonably well with the experimental values. It is found that the external reverberation time in a built-up area depends chiefly on the average building height, and to a lesser extent on the packing fraction, and the ratio of surface area to cross-sectional area of buildings which make up the built-up environment, apart from the absorptive properties of the building and ground plane surfaces. For the spatial distribution of the steady state sound field amplitude in a built-up environment, it is found that the diffuse field amplitude attenuates with distance from a fixed, constant power source exponentially faster than the inverse square law.     

Amplitude modulation in reverberation parametric receiving arrays

The amplitude modulation mechanism in parametric arrays using the backscattering of high-frequency radiation is considered. The effects arising when a reverberation receiving array is used are analyzed.