一种混响背景下的自适应动目标检测方法

本文提出了一种混响和噪声干扰下动目标检测的自适应方法,其结构为自适应混响抵消器（ARC）加推广的自适应相干累积器（GACI）。用自适应混响抵消实现干扰背景的预白化,用推广的自适应相干累积实现匹配滤波的功能。该方法不需要混响干扰的先验知识,也不需要知道信号参量,就可以实现混响干扰背景下动目标的检测。仿真和试验表明,其具有良好的动目标检测性能,且具有很好的海洋信道适应能力。     

多基地空时码探测信号设计及时反相关检测技术

为实现浅海复杂环境下的多基地声纳多源目标回波分辨,本文设计了一种适用于多入多出垂直阵信道环境下的空时码探测信号,并针对倾斜垂直阵的多途子信道差异问题,提出了信号的时反相关检测技术.空时码探测信号采用伪随机信号调制,具有良好的正交性,能在抗子信道严重衰落的同时,分辨多源目标回波.垂直阵受水流冲击,呈倾斜状态时,其多途子信道不一致性会导致各子信道传递信号无法在接收端聚焦,使阵列增益受损,同时导致时延测量能力下降和信号判决错误率上升,为此本文设计了信道训练信号用以估计多途子信道环境,通过虚拟时间反转镜获得子信道不一致条件下的最佳匹配检测信号,实现对接收信号的时反相关检测.仿真结果表明,本文所设计的探测信号和检测方法,能够克服复杂的信道条件和多途子信道不一致性引起的检测问题,满足多基地声纳探测需求,实现多源目标回波分辨.     

基于空时联合稀疏重构的红外小弱运动目标检测方法

提出了一种基于空时联合稀疏重构的红外小弱运动目标检测算法。通过学习序列图像内容而构建的空时联合字典能同时刻画目标或背景的形态特征和运动信息;利用多元高斯运动模式从空时联合字典中提取出目标空时字典和背景空时字典,目标空时过完备字典描述移动的目标,背景空时过完备字典表征背景噪声。将连续多帧图像在空时联合字典上进行稀疏分解,然后分别利用目标空时字典和背景空时字典中的最大稀疏系数及其空时原子重构信号,获取重构残余能量差异来区分目标和背景。试验结果表明,由同源的空时字典重构的残余能量小,而由异构的空时字典恢复的残余能量大,该方法不仅能提高序列信号表示的稀疏度,还能有效提高小运动目标的探测能力。     

基于空时联合稀疏重构的红外小弱运动目标检测方法

提出了一种基于空时联合稀疏重构的红外小弱运动目标检测算法。通过学习序列图像内容而构建的空时联合字典能同时刻画目标或背景的形态特征和运动信息;利用多元高斯运动模式从空时联合字典中提取出目标空时字典和背景空时字典,目标空时过完备字典描述移动的目标,背景空时过完备字典表征背景噪声。将连续多帧图像在空时联合字典上进行稀疏分解,然后分别利用目标空时字典和背景空时字典中的最大稀疏系数及其空时原子重构信号,获取重构残余能量差异来区分目标和背景。试验结果表明,由同源的空时字典重构的残余能量小,而由异构的空时字典恢复的残余能量大,该方法不仅能提高序列信号表示的稀疏度,还能有效提高小运动目标的探测能力。     

海洋混响信号的自适应非线性预测

基于非线性动力学对海洋混响建模,提出了一种自适应非线性预测方法,通过设计一个滤波器来逼近混响信号在相空间的演化轨迹;采用在不同海域进行的两次海上实验获取的混响信号,对自适应预测方法进行验证,其单步预测结果与实验信号符合较好;以模型的单步预测误差作为检验统计量来检测混响中的目标回波,可以在一定程度上抑制混响干扰,提高信混比。     

一种适用于多输入多输出声呐的稳健空时自适应检测方法

针对多输入多输出声呐在水下探测场景中获得的空时快拍数量不足导致空时自适应检测性能下降的问题, 提出一种适用于小空时快拍数情况下的稳健空时自适应检测(STAD)方法。首先将斜对称特性推广到多输入多输出(MIMO)声呐空时自适应检测中, 并利用该特性将空时快拍数量扩展至原来的两倍; 然后利用迭代自适应方法实现对空时协方差矩阵的重构; 最后根据广义似然比检测(GLRT)准则推导出检验统计量。计算机仿真结果表明, 所提方法可以在小空时快拍数情况下准确估计空时协方差矩阵, 大幅减小空时自适应检测对空时快拍数量的需求, 提高多输入多输出声呐在小空时快拍数情况下的检测稳健性。     

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。     

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.     

一种地下爆破震动信号检测及处理系统的研究

采用软硬件结合的设计手段,设计了一种用于地下爆破震动信号检测及处理的系统;系统硬件部分由传感器、存储器、USB等部分组成,负责采集原始震动信号;软件部分是以LabWindows/CVI为平台的虚拟示波器,不仅具有传统示波器的功能,还能完成对震动信号不同方向轴的分量进行合成、滤波、提取震动信号初至波时间等数据处理与分析;实验结果证明,在20 m*20 m*3 m浅层区域内,系统能够无失真的获取地下多维震动矢量信号,并能够对震动信号进行处理显示,达到了预期的目标。     

基于模型更新与快速重检测的长时目标跟踪

在目标遮挡、光线变化等复杂的跟踪环境下,现有相关滤波跟踪算法无法对目标进行长时间实时稳定跟踪。提出一种基于模型更新与快速重检测的长时跟踪算法。首先,在现有的目标定位与尺度变化的相关滤波跟踪算法基础上搭建长时目标跟踪的框架,提出加入模型监测更新机制,根据最大响应和平均峰响应相关能量值判别进入更新或重检测环节;然后,基于提取描述子特征的重检测方法,将提取特征的比特维数统一降到512进行优化,加快重检测速率。所提算法选取OTB-100中20个有代表性的序列进行测试,成功率评估均值为0.706,精确度评估均值为0.805,平均速度为48.5 frame/s;在自采集的数据集上平均准确率能达到87.65%,能够在尺度变化、遮挡等复杂情况下满足长时跟踪的准确性和实时性要求。     

Reverberation time measuring system with correlation technique

It is very difficult to maintain the signal to noise ratio when one is measuring reverberation time at a site or in a high ambient noise field. The application of a correlation technique to such measurements is considered in this paper. The method allows the reverberation time to be measured without subjective interactions. A comparison with the method normally used shows that the method described herein shows a high degree of reliability.     

基于特征分析的自适应干扰抑制

针对水声环境中真实目标常常被强干扰掩盖而无法识别的问题,提出了一种基于特征分析的自适应干扰抑制方法(EAAIS)。根据目标信号可能的方向范围,通过构造合适的判决因子来自适应地估计并抑制互谱密度矩阵特征空间中的非目标信号子空间,得到具有更高信干比和信噪比的目标信号,为更好地实现目标识别与跟踪提供技术基础。数值仿真和海试数据验证结果表明,EAAIS方法能够自适应地抑制目标信号可能方向范围之外的强干扰,显著提高接收目标信号的信噪比和信干比,并获得更可靠的目标方位参数。相对于其它干扰抑制方法,本文提出的方法表现出了更为稳健的干扰抑制能力,具有更宽的适用条件。     

Reverberation time measurements in non-diffuse acoustic field by the modal reverberation time

The increasing presence of low frequency sources and the lack of acoustic standard measurement procedures make the extension of reverberation time measurements to frequencies below 100 Hz necessary. In typical ordinary rooms with volumes between 30 m³ and 200 m³ the sound field is non-diffuse at such low frequencies, entailing inhomogeneities in space and frequency domains. Presence of standing waves is also the main cause of bad quality of listening in terms of clarity and rumble effects. Since standard measurements according to ISO 3382 fail to achieve accurate and precise values in third octave bands due to non-linear decays caused by room modes, a new approach based on reverberation time measurements of single resonant frequencies (the modal reverberation time) has been introduced. From background theory, due to the intrinsic relation between modal decays and half bandwidth of resonant frequencies, two measurement methods have been proposed together with proper measurement procedures: a direct method based on interrupted source signal method, and an indirect method based on half bandwidth measurements. With microphones placed at corners of rectangular rooms in order to detect all modes and maximize SNRs, different source signals were tested. Anti-resonant sine waves and sweep signal turned out to be the most suitable for direct and indirect measurement methods respectively. From spatial measurements in an empty rectangular test room, comparison between direct and indirect methods showed good and significant agreements. This is the first experimental validation of the relation between resonant half bandwidth and modal reverberation time. Furthermore, comparisons between means and standard deviations of modal reverberation times and standard reverberation times in third octave bands confirm the inadequacy of standard procedure to get accurate and precise values at low frequencies with respect to the modal approach. Modal reverberation time measurements applied to furnished ordinary rooms confirm previous results in the limit of modal sound field: for highly damped modes due to furniture or acoustic treatment, the indirect method is not applicable due to strong suppression of modes and the consequent deviation of the acoustic field from a non-diffuse condition to a damped modal condition, while standard reverberation times align with direct method values. In the future, further investigations will be necessary in different rooms to improve uncertainty evaluation.     

Generating Artificial Reverberation via Genetic Algorithms for Real-Time Applications

We introduce a Virtual Studio Technology (VST) 2 audio effect plugin that performs convolution reverb using synthetic Room Impulse Responses (RIRs) generated via a Genetic Algorithm (GA). The parameters of the plugin include some of those defined under the ISO 3382-1 standard (e.g., reverberation time, early decay time, and clarity), which are used to determine the fitness values of potential RIRs so that the user has some control over the shape of the resulting RIRs. In the GA, these RIRs are initially generated via a custom Gaussian noise method, and then evolve via truncation selection, random weighted average crossover, and mutation via Gaussian multiplication in order to produce RIRs that resemble real-world, recorded ones. Binaural Room Impulse Responses (BRIRs) can also be generated by assigning two different RIRs to the left and right stereo channels. With the proposed audio effect, new RIRs that represent virtual rooms, some of which may even be impossible to replicate in the physical world, can be generated and stored. Objective evaluation of the GA shows that contradictory combinations of parameter values will produce RIRs with low fitness. Additionally, through subjective evaluation, it was determined that RIRs generated by the GA were still perceptually distinguishable from similar real-world RIRs, but the perceptual differences were reduced when longer execution times were used for generating the RIRs or the unprocessed audio signals were comprised of only speech.     

Reverberation time measurement by the product of two room impulse responses

The decay function for the evaluation of the reverberation time is often obtained by the method of the backward integration of a squared room impulse response as suggested by M.R. Schroeder more than four decades ago. Since then much work has been published about its implementation. However, soon after the initial exploitation of the method, it was realized that the effects of the background noise contaminating the room impulse response required a careful consideration for accomplishing better results.This paper describes an alternative method dealing with the problem of the backward integration of noisy room impulse responses. This method is based on the processing of two impulse responses sequentially recorded for a fixed source and receiver arrangement in a room. Statistical criteria are proposed to identify how the effect of the noise corrupts the level decay curve using a noise-free synthesized room impulse response as well as measurements performed in a real room.     

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统一转换至速度域,设定合适的速度门限,去除静止的混响成分。通过仿真分析了 SFM信号的混响抑制能力、调制指数和带宽的关系,在设定的速度范围内平均混响抑制能力较连续波(CW)信号可提高7.5 dB。动目标试验数据的处理结果表明了该方法能够抑制混响中强杂波造成的虚警,通过速度差异性可有效提高对动目标的检测能力。     

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统一转换至速度域,设定合适的速度门限,去除静止的混响成分。通过仿真分析了 SFM信号的混响抑制能力、调制指数和带宽的关系,在设定的速度范围内平均混响抑制能力较连续波(CW)信号可提高7.5 dB。动目标试验数据的处理结果表明了该方法能够抑制混响中强杂波造成的虚警,通过速度差异性可有效提高对动目标的检测能力。     

On saturation suppression in adaptive vibration control

An anti-saturation scheme is presented for adaptive vibration control with LMS controllers. A new recursive formula for weight updating is derived from the optimization of certain object functions, which is an extension of the normalized LMS algorithm. The anti-saturation formula is capable of suppressing controller output in case of a large disturbance and expected to improve the performance of the controller. Numerical simulation and experiment have been conducted to demonstrate the performance of the anti-saturation scheme and the results have shown that the scheme is effective in alleviating output saturation and as a result improves performance of the controller in vibration suppression.     

Astronomical adaptive optics in parameter space

We present a general, undiscretized formulation of astronomical adaptive optics that encompasses arbitrary guide star sources and deformable mirror configurations. It is shown that wave-front measurements can be represented as samples of an integral transform of the turbulence perturbation and also that the desired information for adaptive correction is a subset of this transformed space. Some properties of this space are explored, and their implications for adaptive optics are discussed.