LMS算法和RLS算法在水声信道通信系统中应用的比较

为了减小水声通信系统中存在的由于多径传播效应引起的符号间干扰,可在系统中使用线性均衡器。首先介绍了LMS算法和RLS算法基本原理,利用Matlab仿真软件,给出了两种算法在水声信道通信系统中的自适应均衡的仿真系统,并对这两种算法对于均衡器的影响进行了比较,最后对比较的结果进行了分析。     

LMS算法和RLS算法在水声信道通信系统中应用的比较

为了减小水声通信系统中存在的由于多径传播效应引起的符号间干扰,可在系统中使用线性均衡器。首先介绍了LMS算法和RLS算法基本原理,利用Matlab仿真软件,给出了两种算法在水声信道通信系统中的自适应均衡的仿真系统,并对这两种算法对于均衡器的影响进行了比较,最后对比较的结果进行了分析。     

时域均衡在OFDM水声通信系统中的应用

为了对抗水声信道时延扩展大于循环前缀长度时引起的码间串扰问题,在基本的OFDM水声通信系统接收端进行DFT变换前加入相对较短的时域均衡器来限制信道冲激响应的长度。分析了基于MMSE准则的时域均衡算法的实现过程以及算法的复杂程度,对该算法在水声环境下的性能进行了仿真分析,并比较了影响系统性能的因素。结果表明通过加入时域均衡器能够有效对抗信道严重的时间离散性,改善OFDM水声通信系统在循环前缀不足时的系统性能。     

一种用于多径衰落水声信道的盲均衡算法研究

信道均衡是现今高速水声通信系统中克服码间干扰一项关键技术,如何提高均衡器的性能是目前研究的热点课题。针对水声信道的时变多普勒特性,在分析信道模型及盲均衡器结构特点的基础上,提出了一种时域解相关的变步长恒模算法,并对算法进行了仿真及试验分析。结果表明改进算法能够有效克服实际水声通信中的码间干扰效应,具有更快的收敛速度和更小的剩余误差,提高了浅海水声通信的有效性和可靠性,具有广阔的工程应用前景。     

一种用于水声通信的快速自适应均衡器设计

信道均衡是现代水声通信系统中克服码间干扰的重要手段,根据时变水声信道需进行信道均衡的要求,设计了一种判决反馈盲均衡器。针对恒模算法在固定步长下存在收敛速度与剩余误差的矛盾缺陷,提出了一种基于剩余误差的变步长恒模算法,并对改进算法进行了计算机仿真及试验测试,结果表明,改进算法无论是在收敛性能还是在均衡效果上都有大幅提高,在实际水声通信中具有较好的应用价值。     

稀疏水声信道判决反馈盲均衡算法研究

针对高速水声通信中信道的稀疏特性,提出了一种基于常数模准则的稀疏水声信道判决反馈盲均衡算法。该算法将改进的常数模算法与一种变化的判决反馈均衡器结构(部分反馈均衡器)有机结合,利用水声信道的稀疏特性,不但很好地实现了稀疏水声信道的盲均衡,而且简化了计算,易于算法的硬件实现。用典型稀疏水声信道进行了计算机仿真。结果表明,该算法性能稳定,计算量小,稳态均方误差低,整体性能与基于自适应LMS的稀疏迭代算法接近。该研究为高速水声通信中稀疏信道的均衡提供了一种可实现的方法。     

水声通信中基于软判决的块迭代判决反馈均衡器

在单载波频域均衡水声通信系统中,混合结构的时-频域判决反馈均衡器(H-DFE)计算复杂度高,不利于实时实现;而基于硬判决的块迭代判决反馈均衡器(HD-IBDFE)存在错误符号判决造成系统性能下降问题,同时需要估计判决符号和发射数据之间的互相关函数。该文对水声通信中基于软判决的块迭代判决反馈均衡(SD-IBDFE)接收机算法进行了研究,通过对均衡器输出信号进行软判决,并将符号软信息进行反馈,提高了系统性能,同时采用迭代信道估计方法来适应水声信道的时变性。通过仿真比较得出,该方法在水声信道条件下明显优于HD- IBDFE。对湖上试验数据处理结果表明,在浅水1.8 km通信距离下,单通道无编码QPSK调制可实现10-3的误码率并达到3000 bps的有效数据率。     

全海深高速水声通信机设计与试验

为了适应深海科学考察领域的快速发展,实现对全海域水声通信能力的覆盖,本文设计了一种全海深水声通信机。该水声通信机采用大功率D类功率放大器进行信号发射,利用三基元接收通道进行空间分集接收,最大通信距离12 km。水声通信机采用单载波相干水声通信技术,相干解调算法采用了时频域双向turbo均衡器结构,均衡器系数调整采用了改进比例归一化最小误符号率准则算法。该通信机于2018年12月12日在海洋最深处马里亚纳海沟挑战者深渊进行了实时水声通信试验,实现了水下10500 m到水面的高速水声通信,通信速率6000 bps,通信误比特率为10-5量级。      

通信

TN912011011101时域均衡在OFDM水声通信系统中的应用/王熹,吴长奇(燕山大学信息科学与工程学院)//无线电通信技术.―2010,36(5).―55～58.为了对抗水声信道时延扩展大于循环前缀长度时引起的码间串扰问题,在基本的OFDM水声通信系统接收端进行DFT变换前加入相对较短的时域均衡器来限制信道冲激响应的长度。分析了基于MMSE准则的时域均衡算法的实现过程以及算法的复杂程度,对该算法在水声环境下的性能     

一种快速收敛的水声信道分数间隔盲均衡算法

为了加快常数模盲均衡算法(CMA)的收敛速度并降低剩余均方误差,提出了一种快速收敛的分数间隔常数模算法。新算法采用T/2分数间隔均衡器,针对水声通信中常用的BPSK信号,设计了一个新的误差函数,取代常数模算法中的误差函数。最后,采用浅海水声信道进行了仿真,结果表明,新算法性能稳定;收敛速度高于常数模算法和分数间隔常数模算法;剩余均方误差与分数间隔常数模算法处于同一水平。     

Resolution Improvement of Scanning Acoustic Microscopy Using Sparse Signal Representation

Scanning acoustic microscopy is an imaging method in which the focused high frequency ultrasound is used to visualize the micro structures. The morphology and acoustic properties of the biological tissues can be evaluated using scanning acoustic microscope system. To determine thin tissues having micrometer thickness, the high acoustic frequency is required for conventional SAM. In practice the acoustic frequency is restricted by the penetration depth through the material. Characterization of thin sliced tissue is difficult, as the reflected signals from top and bottom are superimposed. In order to improve the axial resolution of conventional SAM, a technique based on sparse signal representation in overcomplete time–frequency dictionaries is investigated and among the great number of algorithms for finding sparse representation, we first apply matching pursuit (MP) and basis pursuit (BP) and then propose the orthogonal matching pursuit (OMP) and stagewise orthogonal matching pursuit (StOMP) algorithms to decompose the A-scan signal to an overcomplete Gabor dictionary. Different criteria are used for measuring the performance of these algorithms in C-scan imaging. The proposed method can separate closely space overlapping echoes beyond the resolution of conventional SAM systems and also the final results show that StOMP performs best overall in extracting the specific echo, since this algorithm is precise and fast.     

Accurate Localization Under Correlated Errors for Underwater Acoustic Sensor Networks

Localization is an indispensable part for most underwater wireless sensor networks (UWSNs). In these networks, acoustic ranging signals are typically used. Because of the depth dependent sound speed, acoustic ray bends in water. This presents a unique problem making conventional localization algorithms assuming straight line distances between target and anchor nodes inaccurate. In this paper, authors present a Maximum Likelihood estimation scheme for accurate location estimation in UWSNs where there is correlation of range measurement errors. The method takes into account ray bending by making use of ray tracing, with sound speed profile as input. The performance of the scheme was studied by means of Monte-Carlo simulation and the results are reported. The Cramer-Rao lower bound of the scheme is also derived and reported.     

Identification of certain time-varying nonlinear Wiener andHammerstein systems

The problem of identification and tracking of time-varying nonlinear systems is addressed. In particular, the Wiener system that consists of a dynamic time-varying linear part followed by a fixed nonlinearity and the Hammerstein system in which the order of these two blocks is reversed are studied. The extended Kalman filter (EKF) algorithm is applied. It is also shown that this algorithm can be reformulated in terms of a nonlinear minimization problem with a quadratic inequality constraint in order to ensure exponential stability, resulting in the algorithm CEKF. As indicated by means of numerical examples, this latter algorithm is less sensitive to the chosen initialization than the EKF. The proposed algorithms depend on certain second-order statistics that may be unknown in a typical scenario. A method for estimation of these quantities is proposed. It is demonstrated that the suggested algorithms can be successfully applied to the problem of acoustic echo cancelation     

激光空中探测水下目标的新方法研究

为了实现在空中利用激光技术对水下目标进行探测的目的,基于迈克尔逊干涉仪的基本原理,提出了一种从水面散射激光中获取水下声源信息的新方法。设计并实现了一套空中探测水下声信号的实验装置。实验结果表明：频谱分析图中的峰值频率即为水下声源的发声频率,系统能够实时探测发声频率在1 kHz～14 kHz的水下声源,且测量标准偏差小于7 Hz。该方法为航空遥感水下目标提供了一条新的技术途径。     

基于支持向量机的水声信道盲均衡算法研究

水声信道的带宽极为有限,与需要训练序列的自适应均衡算法相比盲均衡技术节省了带宽,特别适合高速水声通信和多点通信。针对基于高阶统计量的盲均衡算法收敛速度慢的缺点,研究了一种基于支持向量机的盲均衡算法,它收敛速度快并具有全局最优解。通过对浅海信道触发通信信号的计算机仿真,证明了算法的有效性。     

Artifact Trapping During Time Reversal Photoacoustic Imaging for Acoustically Heterogeneous Media

Several different reconstruction algorithms have been proposed for photoacoustic tomography, most of which presuppose that the acoustic properties of the medium are constant and homogeneous. In practice, there are often unknown spatial variations in the acoustic properties, and these algorithms give, at best, only approximate estimates of the true image. The question as to which approach is the most robust in these circumstances is therefore one of practical importance. Image reconstruction by “time reversal”—using a numerical propagation model with a time-varying boundary condition corresponding to the measured data in reversed temporal order—has been shown to be less restrictive in its assumptions than most, and therefore a good candidate for a general and practically useful algorithm. Here, it is shown that such reconstruction algorithms can “trap” time reversed scattered waves, leading to artifacts within the image region. Two ways to mitigate this effect are proposed.      

Acoustic feedback cancellation for long acoustic paths using a nonstationary source model

While several proactive acoustic feedback (Larsen-effect) cancellation schemes have been presented for speech applications with short acoustic feedback paths as encountered in hearing aids, these schemes fail with the long impulse responses inherent to, for instance, public address systems. We derive a new prediction error method (PEM)-based scheme (referred to as PEM-AFROW) which identifies both the acoustic feedback path and the nonstationary speech source model. A cascade of a short- and a long-term predictor removes the coloring and periodicity in voiced speech segments, which account for the unwanted correlation between the loudspeaker signal and the speech source signal. The predictors calculate row operations which are applied to prewhiten the speech source signal, resulting in a least squares system that is solved recursively by means of normalized least mean square or recursive least squares algorithms. Simulations show that this approach is indeed superior to earlier approaches whenever long acoustic channels are dealt with.