基于R阵动态估计的自适应滤波算法

针对组合导航系统中观测噪声特性复杂多变、难于准确估计的问题,基于不同测量系统的测量互补特性,提出了针对单次历元的观测噪声特性动态估计方法。在此基础上,以预设滤波精度为指标,提出了通过构造自适应因子对估计观测噪声进行适当调节的自适应卡尔曼滤波算法。该算法通过构造相对测量关系,避免了直接对测量噪声真值求解的难题,并且在滤波过程中采用序贯处理方法进行实时解算,有效降低了计算量。在GPS/DR实际系统中的应用结果表明,同改进的sage-husa算法及MAKF算法相比,基于R阵动态估计的自适应滤波算法能够自适应地跟踪GPS测量噪声特性的变化,定位结果光滑可靠,具有明显的优越性。     

一种抗冲击噪声的对数总体最小二乘自适应滤波算法

在未知系统输入信号和输出信号均含有噪声的环境中,传统的自适应滤波算法,如最小均方(LMS)算法,会产生有偏估计.总体最小二乘(TLS)算法能够同时最小化输入信号与输出信号的噪声干扰,是解决此类问题的重要方法.然而,在许多实际应用中,干扰噪声可能具有冲击特性,这使得传统基于2阶统计量的自适应滤波算法,包括总体最小二乘算法...     

Frequency tracking techniques of power systems in coloured noises

A technique for tracking the frequency of power systems in sine noises using numerical differentiation is presented. A voltage or current sinusoidal signal corrupted by sine noises and white noises is considered. For the signal corrupted by one or two sine noises, a central numerical differentiation-based method is proposed. For the signal corrupted by multiple sine noises and white noises, a hybrid method of numerical differentiation and a digital finite impulse response (FIR) filter are proposed. The digital FIR filtering algorithm is used to remove the white noises and the sine noises, and the numerical differentiation algorithm is used to estimate the fundamental frequency of power systems when the fundamental component is decomposed out of the signal. The proposed algorithm shows an advantage in time and speed when compared with other existing techniques and shows better dynamics and higher accuracy in frequency estimation. Carried out in Matlab, the simulation results are satisfactory     

CO-OFDM系统中一种新颖相位噪声抑制算法

针对相干光正交频分复用(CO-OFDM)系统中相位噪声造成的严重影响,提出了一种采用线性插值和卡尔曼滤波相结合的相位噪声抑制算法.该算法在第一阶中对接收端的时域信号进行线性组合,之后利用线性插值算法对相位噪声进行初步抑制,在第二阶中采用卡尔曼滤波技术来抑制残余的相位噪声.分析和仿真表明:提出的二阶算法能够有效地抑制相位噪声对OFDM符号的影响,在相位噪声线宽较大时明显地降低错误平层,利用所提出的二阶算法可使误码率达到10-7以下,有效地提高了系统的性能.     

Sparse Least Logarithmic Absolute Difference Algorithm with Correntropy-Induced Metric Penalty

Sparse adaptive filtering algorithms are utilized to exploit system sparsity as well as to mitigate interferences in many applications such as channel estimation and system identification. In order to improve the robustness of the sparse adaptive filtering, a novel adaptive filter is developed in this work by incorporating a correntropy-induced metric (CIM) constraint into the least logarithmic absolute difference (LLAD) algorithm. The CIM as an \(l_{0}\)-norm approximation exerts a zero attraction, and hence, the LLAD algorithm performs well with robustness against impulsive noises. Numerical simulation results show that the proposed algorithm may achieve much better performance than other robust and sparse adaptive filtering algorithms such as the least mean p-power algorithm with \(l_{1}\)-norm or reweighted \(l_{1}\)-norm constraints.     

α噪声环境下整体最小平均P-范数IIR自适应滤波算法

当无限冲激响应(IIR)系统输入和输出信号中都存在α稳定噪声干扰,传统的最小平均P-范数算法(LMP)的解会出现较大偏差,本文提出了一种自适应IIR滤波整体最小平均P-范数(IIR_TLMP)算法,算法中整体考虑输入和输出信号的α稳定噪声干扰,将最小化l_p范数Rayleigh商采用随机梯度法得到自适应IIR滤波方程。通过仿真首先考察了特征指数和步长因子等主要参数对TLMP算法性能的影响,最后分别在时不变和时变系统中,将TLMP算法与LMP算法的性能在进行了比较,结果显示TLMP有更快的收敛速度和更小的误差。      

基于新息的自适应增量Kalman滤波器

在一定环境条件下,当系统的量测方程没有进行验证或校准时,使用该量测方程往往会产生未知的系统误差,从而导致较大的滤波误差。增量方程的引入可以有效解决欠观测系统的状态估计问题。该文考虑带未知噪声统计的线性离散增量系统,首先提出一种基于新息的噪声统计估计算法。可以得到系统噪声统计的无偏估计。进而,提出一种新的增量系统自适应Kalman滤波算法。相比已有的自适应增量滤波算法,该文所提算法得到的状态估计精度更高。两个仿真实例证明了其有效性和可行性。     

带相关噪声的观测融合稳态Kalman滤波算法及其全局最优性

对于带相关的输入白噪声和观测白噪声及相关观测白噪声的多传感器线性离散定常随机系统,用加权最小二乘(WLS)法提出了一种加权观测融合稳态Kalman滤波算法,可处理状态、白噪声和信号融合滤波、平滑、预报问题。基于稳态信息滤波器证明了它完全功能等价于集中式观测融合稳态Kalman滤波算法,因而它具有渐近全局最优性,且可减少计算负担。一个跟踪系统仿真例子验证了它的功能等价性。     

一种改进的变步长LMS自适应算法

为了提高LMS自适应算法的性能,在对一些变步长LMS算法分析的基础上,提出了步长因子μ(n)与误差信号e(n)之间一种新的非线性函数关系,该算法比固定步长LMS算法收敛速度快,稳定性好,另外与文[5]中算法相比,不需要进行指数运算.将该算法应用于自适应噪声抵消系统的仿真中,计算机仿真结果与实际分析相一致.     

An improved proportionate least mean <Emphasis Type="Italic">p</Emphasis>-power algorithm for adaptive filtering

The least mean p-power error criterion has been successfully used in adaptive filtering due to its strong robustness against large outliers. In this paper, we develop a new adaptive filtering algorithm, named the proportionate least mean p-power (PLMP) algorithm, which uses the mean p-power error as the adaptation cost function. Compared with the standard proportionate normalized least mean square algorithm, the PLMP can achieve much better performance in terms of the mean square deviation, especially in the presence of impulsive non-Gaussian noises. The mean and mean square convergence of the proposed algorithm are analyzed, and some related theoretical results are also obtained. Simulation results are presented to verify the effectiveness of our proposed algorithm.     

基于最小离差准则的自适应滤波器设计

实际系统中存在大量的具有脉冲性质的非高斯噪声，本文介绍利用实数维低阶矩理论，根据最小离差准则设计自适应滤波器，去除一类服从稳定分布的噪声，并给出了基于LMAD算法的自适应滤波器．实验表明，这种滤波器去除SaS分布噪声的性能比基于LMS算法的滤波器优越。     

降秩自适应滤波算法研究

对降秩自适应滤波算法进行了系统的总结和分析,推导了其相互关系。分析表明,GSC（Generalized Sidelobe Canceller）框架降秩变换自适应滤波是各种降秩自适应滤波算法的统一模型。在此基础上导出了线性约束正交投影算法。降秩多级维纳滤波器在相关意义上进行截断降秩,其降秩性能优于基于特征子空间截断的降秩方法。酉多级维纳滤波器与共轭梯度法等效,均是基于Krylov子空间截断降秩的方法,降秩性能更优。最后通过计算机仿真试验比较了各种降秩处理算法的性能。     

Narrow-band interference excision in spread-spectrum systems usingself-orthogonalizing transform-domain adaptive filters

Among many transform-domain interference excision techniques, transform-domain adaptive filtering has many advantages. It is based on a true optimization of some particular performance parameters such as the bit-error rate (BER). Moreover, it is insensitive to jammer frequency. However, transform-domain adaptive filtering also has the drawback of being incapable of tracking a rapidly changing interference because most adaptive algorithms require time to converge to the optimal solution. In this paper, a self-orthogonalizing transform-domain least mean square (SO-TRLMS) algorithm is used to speed up the convergence. Compared to a traditional transform-domain least mean square (TRLMS) algorithm, the SO-TRLMS algorithm can significantly improve the convergence rate of the LMS algorithm, thus making the transform-domain adaptive filtering technique more suitable for real-time processing. In order to show how the system performance is affected by various factors such as interference power and the transform used, this paper presents an analytical result for the BER performance that is applicable for arbitrary orthogonal linear transforms. Simulation results are also presented to demonstrate the validity of the analysis     

两种改进型中值滤波算法比较

重点介绍了综合型中值滤波算法以及一种自适应中值滤波算法．针对这两种改进型中值滤波算法,对含有高斯噪声,椒盐噪声,混合噪声以及高密度噪声图像进行去噪处理,比较综合型中值滤波算法（文中采用了十字型和交叉型）和自适应中值滤波算法对不同图像的去噪效果．     

Second-order adaptive Volterra system identification based ondiscrete nonlinear Wiener model

The authors present the nonlinear LMS adaptive filtering algorithm based on the discrete nonlinear Wiener (1942) model for second-order Volterra system identification application. The main approach is to perform a complete orthogonalisation procedure on the truncated Volterra series. This allows the use of the LMS adaptive linear filtering algorithm for calculating all the coefficients with efficiency. This orthogonalisation method is based on the nonlinear discrete Wiener model. It contains three sections: a single-input multi-output linear with memory section, a multi-input, multi-output nonlinear no-memory section and a multi-input, single-output amplification and summary section. For a white Gaussian noise input signal, the autocorrelation matrix of the adaptive filter input vector can be diagonalised unlike when using the Volterra model. This dramatically reduces the eigenvalue spread and results in more rapid convergence. Also, the discrete nonlinear Wiener model adaptive system allows us to represent a complicated Volterra system with only few coefficient terms. In general, it can also identify the nonlinear system without over-parameterisation. A theoretical performance analysis of steady-state behaviour is presented. Computer simulations are also included to verify the theory     

基于经验模态分解的激光陀螺随机信号消噪

各种随机噪声是导致激光陀螺产生误差的主要因素,且其性质特殊,很难用传统的滤波方法去除。为了抑制激光陀螺的随机漂移,提高使用精度,提出了一种新型经验模态分解方法对陀螺随机漂移测试信号进行滤波处理。该方法将经验模态分解的内模函数中两个相邻过零点之间的信号定义为模态单元,并作为基本分析对象,通过对模态单元振幅的阈值处理来判断模态单元的类型,进而建立模态单元滤波模型。分析了经验模态分解法在分解不同Hurst指数分形高斯噪声时模态振幅的演化规律,并建立了一种用于高斯消噪的阈值选取规则。运用该方法对激光陀螺测试数据进行了滤波降噪实验,并用Allan方差法对不同降噪算法的降噪效果进行了比较分析,实验结果验证了该方法的有效性和优越性。     

A Novel Normalized Sign Algorithm for System Identification Under Impulsive Noise Interference

To overcome the performance degradation of adaptive filtering algorithms in the presence of impulsive noise, a novel normalized sign algorithm (NSA) based on a convex combination strategy, called NSA-NSA, is proposed in this paper. The proposed algorithm is capable of solving the conflicting requirement of fast convergence rate and low steady-state error for an individual NSA filter. To further improve the robustness to impulsive noises, a mixing parameter updating formula based on a sign cost function is derived. Moreover, a tracking weight transfer scheme of coefficients from a fast NSA filter to a slow NSA filter is proposed to speed up the convergence rate. The convergence behavior and performance of the new algorithm are verified by theoretical analysis and simulation studies.     

Robust Kalman filters for linear time-varying systems withstochastic parametric uncertainties

We present a robust recursive Kalman filtering algorithm that addresses estimation problems that arise in linear time-varying systems with stochastic parametric uncertainties. The filter has a one-step predictor-corrector structure and minimizes an upper bound of the mean square estimation error at each step, with the minimization reduced to a convex optimization problem based on linear matrix inequalities. The algorithm is shown to converge when the system is mean square stable and the state space matrices are time invariant. A numerical example consisting of equalizer design for a communication channel demonstrates that our algorithm offers considerable improvement in performance when compared with conventional Kalman filtering techniques     

最大互相关熵多凸组合自适应滤波算法

基于最大互相关熵准则(MCC)的自适应滤波算法在非高斯噪声环境下具有强鲁棒性,得到了广泛应用.然而,传统MCC滤波算法在选择参数时依然受到收敛速度与稳态精度之间固有矛盾的困扰.为解决这一问题,该文提出一类多凸组合MCC算法,能够充分发挥不同参数组合下滤波算法的性能优势,从而获得更好的信道跟踪能力.理论分析得出了所提算法...