一种变正则化矩阵的改进多带结构子带自适应滤波算法

定正则化因子的改进多带结构子带自适应滤波(IMSAF)算法在取得收敛速度快和稳态失调误差小之间存在冲突.根据系统噪声抵消原理,设定子带后验误差功率等于子带噪声功率,本文提出了变正则化矩阵的IMSAF算法来解决这一问题.仿真结果证明,所提算法可以同时达到收敛速度快、稳态失调误差小以及追踪速度快等优势.     

基于小波变换的变步长LMS自适应滤波算法

变步长LMS自适应滤波算法通过构造合适的步长因子有效的解决了传统LMS算法收敛速度和稳态误差相矛盾的问题.变换域LMS自适应滤波算法通过正交变换降低了输入信号矩阵的相关性,提高了算法的收敛速度.将这两种算法相结合,提出了一种新的基于小波变换的变步长LMS自适应滤波算法.仿真结果表明,该算法无论是收敛速度还是稳态误差都有了很大的提高.     

自适应噪声对消的归一化LMS算法

为了克服自适应滤波中固定步长LMS算法存在收敛速度与稳态误差的矛盾,本文通过MATLAB仿真不同步长因子下LMS算法的学习曲线,分析了LMS算法在收敛过程中存在的矛盾,并运用归一化LMS(NLMS)算法来改善上述矛盾。NLMS算法是通过输入变量改变步长因子从而改变算法的收敛特性。本文对NLMS与LMS算法的误差曲线仿真并进行稳态误差效果比较,结果显示NLMS算法的稳态误差精确度明显提高,收敛速度加快。通过将LMS算法与NLMS算法应用于自适应噪声对消中,得到NLMS算法具有收敛速度更快同时稳态误差更小的特性,该算法能够快速对干扰信号作出反应,使除噪效果更好。     

α稳定分布下的自适应数据块滤波算法

本文基于分数低阶统计量原理提出了α稳定分布下的自适应数据块LMP和LMAD滤波算法.新算法利用更多的输入信号和误差信号的信息,更准确地估计梯度,调整自适应滤波器权向量增量方向,提高了收敛速度.采用变步长因子改善算法的稳态误差,进一步提高了算法的收敛性能.仿真结果表明,在分数低阶α稳定分布信号条件下,自适应数据块滤波新算法较NLMP算法和NLMAD算法有更好的收敛性能.     

一种变步长LMS波束形成算法性能研究

智能天线的自适应算法通过迭代运算获取波束形成的最优权值矢量时,收敛速度和稳态误差是衡量一个算法是否优良的关键因素。它们的好坏直接影响着系统波束形成的性能。系统地分析了传统的最小均方（LMs）算法的收敛速度以及稳态误差的性能,在此基础上提出了一种新的变步长LMS算法,将此算法应用于波束形成,并用Matlab软件进行仿真。仿真结果表明,改进后的算法较传统LMS具有较快的收敛速度和较小的稳态误差。     

一种新的变步长变换域自适应滤波算法

变换域LMS算法能通过正交变换有效降低输入信号自相关矩阵特征值的分散程度,可提高算法的收敛速度;变步长LMS算法可以克服固定步长因子所导致的算法在较快收敛速度和较小稳态误差之间存在的矛盾,从而获得较快的收敛速度和较好的收敛结果。将二者相结合,提出了一种新的变步长变换域自适应滤波算法。计算机仿真结果表明该算法具有更快的收敛速度和更小的稳态误差,并且运算量较少,具有良好的实用性能。     

新变步长自适应算法及其时延估计性能

针对已有的变步长自适应算法收敛速度和稳态误差矛盾的问题,提出了一种新的变步长最小均方自适应滤波算法。新的算法在类S函数的基础上,引入调节因子P对步长函数的形状进行实时调整,并以误差的自相关时间均值估计调节步长,使得算法在初始时具有较快的收敛速度,稳态时有更平滑的步长变化。在新算法中引用最大似然加权算法进一步抑制自适应滤波器权系数伪峰。将新算法和最大似然加权应用在自适应时延估计的实验中,结果表明:在已有参数固定的条件下,新提出的算法具有更快的收敛速度和更小的稳态误差。同时,时延估计实验中能有效地实现信噪比-3 dB以上的准确时延估计。     

变步长自适应滤波算法的统一框架及其矢量扩展

针对大量的变步长自适应滤波算法,提出了一种采用约束最优化方法描述变步长自适应滤波算法的统一框架.在该框架下,不同算法的目标函数或决策变量不同.利用该框架,将非参数变步长归一化最小均方误差(NPVSS-NLMS)算法扩展到矢量空间,导出一种新的变步长仿射投影算法.理论分析与计算机仿真表明,该算法不仅能根据输出误差自适应调整步长,而且对强相关输入信号能够保持良好的收敛速度、很小的稳态误差和很快的跟踪速度.将该算法应用于回波抵消,其稳态误差比NPVSS-NLMS算法低近5dB.     

智能天线自适应波束形成算法的研究

智能天线的自适应算法通过迭代运算获取用于波束形成的最优权值矢量时,是否具有较快的收敛速度和较小的稳态误差成为决定波束形成性能的主要因素。据此提出在传统的LMS算法中引入变步长和变换域的思想,采用改进的自：适应算法用于波束形成。MATLAB仿真结果表明,该算法具有较快的收敛速度和较小的稳态误差,波束形成的性能更优。     

改进的变步长变换域最小均方算法

变换域是一种在强相关信号输入时加快自适应算法收敛的方法,但仍然存在收敛速度的要求与稳态失调的要求相矛盾的问题。本文在变换域最小均方误差算法（transform domain LMS, TDLMS）的基础上提出了一种改进的变步长方案,其变步长因子受到误差自相关的控制,消除了不相关的观测噪声的影响。本文分别在平稳和非平稳状态下,对算法的收敛和稳态性能进行理论分析,并给出了最佳的算法参数。仿真设置相同的稳态误差,结果表明本文算法在平稳状态下比固定步长的算法提前1300点收敛,在非平稳状态下提前1400点收敛,且与文献中其它变步长的算法相比收敛速度均有提升。      

基于稀疏和低秩恢复的稳健DOA估计方法

该文针对有限次采样导致传统波达方向角(DOA)估计算法存在较大估计误差的问题,提出一种基于稀疏低秩分解(SLRD)的稳健DOA估计方法。首先,基于低秩矩阵分解方法,将接收信号协方差矩阵建模为低秩无噪协方差及稀疏噪声协方差矩阵之和;而后基于低秩恢复理论,构造关于信号和噪声协方差矩阵的凸优化问题;再者构建关于采样协方差矩阵估计误差的凸模型,并将此凸集显式包含进凸优化问题以改善信号协方差矩阵估计性能进而提高DOA估计精度及稳健性;最后基于所得最优无噪声协方差矩阵,利用最小方差无畸变响应(MVDR)方法实现DOA估计。此外,基于采样协方差矩阵估计误差服从渐进正态分布的统计特性,该文推导了一种误差参数因子选取准则以较好重构无噪声协方差矩阵。数值仿真表明,与传统常规波束形成(CBF)、最小方差无畸变响应(MVDR)、传统多重信号分类(MUSIC)及基于稀疏低秩分解的增强拉格朗日乘子(SLD-ALM)算法相比,有限次采样条件下所提算法具有较高DOA估计精度及较好稳健性能。     

A New Normalized Subband Adaptive Filter Algorithm with Individual Variable Step Sizes

Proposed is a novel variable step size normalized subband adaptive filter algorithm, which assigns an individual step size for each subband by minimizing the mean square of the noise-free a posterior subband error. Furthermore, a noniterative shrinkage method is used to recover the noise-free priori subband error from the noisy subband error signal. Simulation results using the colored input signals have demonstrated that the proposed algorithm not only has better tracking capability than the existing subband adaptive filter algorithms, but also exhibits lower steady-state error.     

Estimation-based synthesis of H∞-optimal adaptive FIR filtersfor filtered-LMS problems

This paper presents a systematic synthesis procedure for H∞-optimal adaptive FIR filters in the context of an active noise cancellation (ANC) problem. An estimation interpretation of the adaptive control problem is introduced first. Based on this interpretation, an H∞ estimation problem is formulated, and its finite horizon prediction (filtering) solution is discussed. The solution minimizes the maximum energy gain from the disturbances to the predicted (filtered) estimation error and serves as the adaptation criterion for the weight vector in the adaptive FIR filter. We refer to this adaptation scheme as estimation-based adaptive filtering (EBAF). We show that the steady-state gain vector in the EBAF algorithm approaches that of the classical (normalized) filtered-X LMS algorithm. The error terms, however, are shown to be different. Thus, these classical algorithms can be considered to be approximations of our algorithm. We examine the performance of the proposed EBAF algorithm (both experimentally and in simulation) in an active noise cancellation problem of a one-dimensional (1-D) acoustic duct for both narrowband and broadband cases. Comparisons to the results from a conventional filtered-LMS (FxLMS) algorithm show faster convergence without compromising steady-state performance and/or robustness of the algorithm to feedback contamination of the reference signal     

Two algorithms for adaptive retrieval of slowly time-varyingmultiple cisoids in noise

Two algorithms for tracking parameters of slowly varying multiple complex sine waves (cisoids) in noise (the multiple frequency tracker and the adaptive notch filter) are described. For high signal-to-noise ratio (SNR), the properties of the algorithms (i.e., stability, noise rejection, and tracking speed) are studied analytically using a linear filter approximation technique. The tradeoff between noise rejection and tracking error for both algorithms is shown to be similar. Different choices of the design variables are discussed, namely (i) minimal mean-square estimation error for random walk modeled frequency variations and (ii) minimal stationary estimation variance subject to a given tracking delay     

Robust M-estimate adaptive filtering

An M-estimate adaptive filter for robust adaptive filtering in impulse noise is proposed. Instead of using the conventional least-square cost function, a new cost function based on an M-estimator is used to suppress the effect of impulse noise on the filter weights. The resulting optimal weight vector is governed by an M-estimate normal equation. A recursive least M-estimate (RLM) adaptive algorithm and a robust threshold estimation method are derived for solving this equation. The mean convergence performance of the proposed algorithm is also analysed using the modified Huber (1981) function (a simple but good approximation to the Hampel's three-parts-redescending M-estimate function) and the contaminated Gaussian noise model. Simulation results show that the proposed RLM algorithm has better performance than other recursive least squares (RLS) like algorithms under either a contaminated Gaussian or alpha-stable noise environment. The initial convergence, steady-state error, robustness to system change and computational complexity are also found to be comparable to the conventional RLS algorithm under Gaussian noise alone     

AKF与EFRLS在动态目标跟踪性能上的比较

卡尔曼滤波是具有递推估计形式的最优滤波,但最优性的获得是在过程噪声和观测噪声统计特性已知的前提下得到的。然而,在大量的动态目标跟踪实际问题中噪声具有不确定性,因而有必要研究在噪声不确定下动态目标的跟踪算法以满足实际问题的需要。文中介绍自适应Kalman滤波对过程噪声方差的估计以及推广的遗忘因子最小二乘法对状态估计的递推公式,并且在平均误差最小准则下通过计算机仿真比较两种方法对动态目标的跟踪性能.仿真结果表明,在不确定噪声下自适应Kalman滤波能够取得比推广的遗忘因子递推最小二乘法更好的跟踪性能。     

Low-complexity adaptive step size constrained constant modulus SG algorithms for adaptive beamforming

In this paper, we propose two low-complexity adaptive step size mechanisms to enhance the performance of stochastic gradient (SG) algorithms for adaptive beamforming. The beamformer is designed according to the constrained constant modulus (CCM) criterion and the proposed mechanisms are employed in the SG algorithm for implementation. A complexity comparison is provided to show their advantages over existing methods, and a sufficient condition for the convergence of the mean weight vector is established. Theoretical expressions of the excess mean-squared error (EMSE), in both the steady-state and tracking cases, are derived based on the energy conservation approach. The effects of multiple access interference (MAI) and additive noise are considered. Simulation experiments are presented for both the stationary and non-stationary scenarios, illustrating that the proposed algorithms achieve superior performance compared with existing methods, and verifying the accuracy of the analyses.     

Analysis of stability and performance of adaptation algorithms with time-invariant gains

Adaptation laws that track parameters of linear regression models are investigated. The considered class of algorithms apply linear time-invariant filtering on the instantaneous gradient vector and includes least mean squares (LMS) as its simplest member. The asymptotic stability and steady-state tracking performance for prediction and smoothing estimators is analyzed for parameter variations described by stochastic processes with time-invariant statistics. The analysis is based on a novel technique that decomposes the inherent feedback of adaptation algorithms into one time-invariant loop and one time-varying loop. The impact of the time-varying feedback on the tracking error covariance can be neglected under certain conditions, and the performance analysis then becomes straightforward. Performance analysis in the presence of a non-negligible time-varying feedback is performed for algorithms that use scalar measurements. Convergence in mean square error (MSE) and the MSE tracking performance is investigated, assuming independent consecutive regression vectors. Closed-form expressions for the tracking MSE are thereafter derived without this independence assumption for a subclass of algorithms applied to finite impulse response (FIR) models with white inputs. This class includes Wiener LMS adaptation.     

基于矩阵补全的二阶统计量重构DOA估计方法

该文针对传统波达方向角(DOA)估计算法在非均匀噪声下角度估计精度差及分辨率低的问题,基于矩阵补全理论,提出一种二阶统计量域下加权L1(MC-WLOSRSS)稀疏重构DOA估计算法。首先,基于矩阵补全方法,引入弹性正则化因子将接收信号协方差矩阵重构为无噪声协方差矩阵;而后在二阶统计量域下通过矩阵求和平均将无噪声协方差矩阵多矢量问题转化为单矢量问题;最后利用稀疏重构加权L1范数实现DOA参数估计。数值仿真表明,与传统MUSIC, IL1-SRACV, L1-SVD子空间算法及稀疏重构加权L1算法相比,所提算法能显著抑制非均匀噪声影响,具有较好DOA估计性能,且在低信噪比条件下,亦具有较高估计精度和分辨力。     

A stochastic gradient adaptive filter with gradient adaptive stepsize

The step size of this adaptive filter is changed according to a gradient descent algorithm designed to reduce the squared estimation error during each iteration. An approximate analysis of the performance of the adaptive filter when its inputs are zero mean, white, and Gaussian noise and the set of optimal coefficients are time varying according to a random-walk model is presented. The algorithm has very good convergence speed and low steady-state misadjustment. The tracking performance of these algorithms in nonstationary environments is relatively insensitive to the choice of the parameters of the adaptive filter and is very close to the best possible performance of the least mean square (LMS) algorithm for a large range of values of the step size of the adaptation algorithm. Several simulation examples demonstrating the good properties of the adaptive filters as well as verifying the analytical results are also presented