应用于自适应格型RLS滤波器的变阶数算法

提出了一种新的变阶数(或抽头长度)算法,并将之应用于变阶数自适应格型递归最小二乘(RLS)滤波器的阶数更新中,讨论了格型滤波器阶数更新时相关参数的调整方法。新算法以分贝的形式比较短滤波器与长滤波器的时平均平方误差,采用自适应的抽头长度步长,能够在滤波器权值未收敛时同时快速更新滤波器长度与权值,且在不同大小噪声条件下都能收敛到最优阶数。理论分析与不同大小噪声条件下的自适应系统辨识仿真结果验证了新算法的有效性。      

应用粒子群优化算法设计级联MZI型光滤波器

提出了一种应用粒子群优化算法（PSO）设计极联马赫一曾德尔干涉仪（MZI）型滤波器的新方案。方案巧妙地结合离散型和连续型PSO,利用连续PSO优化耦合角组合、离散PSO优化相移因子组合,由此得到耦合角和相移因子的优化组合,实现光滤波器的优化设计。给出了PSO方案的设计流程及具体步骤,并设计了级联数为3、阶数为5和信道间隔为0．8nm的梳状滤波器,其输出响应与理论滤波器一致。新方案逼近精度高,同时耦合角限制在（0,π／2）内优化配置。     

自适应递归滤波器的一种间接误差控制新算法

本文提出了适用于自适应递归滤波器的一种间接误差控制新算法,简称为IDEC算法,理论分析和大量计算机模拟表明,采用本算法的自适应递归滤波器具有结构简单、计算量小和收敛性好的优点,是对FIR自适应滤波器和LMS算法的一个有力的扩充。     

自适应滤波算法研究及其Matlab实现

在对自适应滤波器相关理论研究的基础上,重点研究了LMS自适应滤波算法,给出了不同信噪比条件下,LMS算法的Matlab仿真实现的滤波结果,通过分析仿真结果可以看出,在一定信噪比范围内,LMS算法在未知信号与噪声统计特性的条件下可以达到较好的滤波效果.     

一种三阶Volterra自适应滤波算法

采用更高阶Volterra滤波器更好地逼近非线性系统时,Volterra自适应滤波算法的计算复杂度呈幂级数增加。针对此问题,本文提出了一种在α稳定分布噪声背景下基于离散余弦变换(DCT)的三阶Volterra滤波算法。首先将Volterra滤波器的三次项权系数矩阵分解成一组二次项权系数矩阵;然后利用正交变换,将二次项权系数矩阵变换成对角矩阵,从而大大减少了权系数个数,有效降低了算法的计算复杂度;最后将Volterra自适应滤波器输出表示成线性滤波器输出形式,并由此得到权系数自适应调整算法。系统辨识的仿真结果表明,本算法在α稳定分布噪声背景下具有优越的性能。     

变抽头长度LMS自适应滤波算法

该文将自适应滤波器抽头长度与权值调整问题归结为单一的权值调整问题,提出了抽头长度的一般更新公式及新的变抽头长度LMS算法,从理论上分析了其合理性与收敛性。新算法用长滤波器与短滤波器的时平均平方误差估计稳态均方误差,采用了自适应调整的抽头长度步长,可在滤波器权值未收敛时就快速更新抽头长度。论文还证明了目前文献中几种有效的变抽头长度算法也可看作或化为文中抽头长度一般更新公式的特例,理论分析与自适应系统辨识的仿真结果验证了新算法的有效性。     

基于α稳定分布的二阶Volterra变抽头长度自适应滤波算法

传统自适应Volterra滤波器抽头长度固定。当一个被识别系统或被均衡信道的特征未知或时变时,自适应滤波器的抽头长度太长,不仅增加了计算量同时也增加了误差;抽头长度太短则无法满足系统的性能要求。针对这个问题本文提出了一种二阶Volterra变抽头长度自适应滤波算法。先对Volterra滤波器输入信号进行格型滤波处理,实现了二次项信号解耦,减少了二次项的权系数,使线性部分和非线性部分权值具有相同的抽头长度,简化了传统Volterra滤波器的结构;基于最小平均p范数准则,运用分数抽头长度的概念,对滤波器抽头长度进行实时自适应调整,用LMP算法自适应调整权系数。计算机仿真结果表明,在不同信噪比的高斯噪声和 稳定分布噪声背景下, 应用本文算法的自适应信道均衡都具有良好的收敛性能,本文算法能自适应调整到最优抽头长度;验证了算法的有效性。      

基于小波变换与形态学运算的ECG自适应滤波算法

针对ECG信号常用滤波算法存在的缺陷,提出了基于小波变换与形态学运算的自适应滤波新算法。该算法利用形态学滤波器去除基线漂移信号,用小波滤波器去除高频干扰信号,并将这两部分所得到的心电噪声分量作为自适应滤波器的参考输入信号,利用自适应滤波器调整对含噪ECG信号进行滤波处理。最后,经实验验证了本文算法的有效性。     

应用于多通道有源控制的自适应组合逆算法

本文在分析单通道伴随LMS算法的基础上,对组合逆算法进行推导和分析,获得了一种有效多通道自适应的快速逆算法,分析了算法的性能,给出了算法的实现步骤。与同样参数条件下的FxLMS算法的运算量进行的比较结果表明: 该算法大大减少了计算量。     

一种变步长凸组合自适应滤波器及其均方性能分析

在分析凸组合最小均方(CLMS)算法性能的基础上,提出一种新的变步长凸组合最小均方(VSCLMS)算法。该算法采用一种变步长滤波器替代原CLMS滤波器组中的恒值大步长滤波器,使新的自适应滤波器能够在噪声、时变,甚至非平稳的环境下保持良好的随动性能,并在收敛的各个阶段均保持快速且稳定的均方特性。理论推导与仿真分析分别验证了新算法与原CLMS算法相比不仅有更快的收敛速度,而且稳态均方性能与跟踪性能也有所提高。     

LMS自适应滤波器的设计理论研究

主要介绍了自适应滤波器的基本理论思想,具体阐述了自适应滤波器的基本原理、算法及设计方法。首先介绍自适应滤波器的原理,然后对FIR结构滤波器做了详细的阐述,自适应算法是整个系统的核心,重点对LMS算法的设计方法,设计步骤做了分析,最后对LMS算法进行MATLAB仿真,根据性能评价标准和实验结果表明,该自适应滤波器滤波效果优越。     

Potential of Particle Swarm Optimization and Genetic Algorithms for FIR Filter Design

This article studies the performance of two metaheuristics, particle swarm optimization (PSO) and genetic algorithms (GA), for FIR filter design. The two approaches aim to find a solution to a given objective function but employ different strategies and computational effort to do so. PSO is a more recent heuristic search method than GA; its dynamics exploit the collaborative behavior of biological populations. Some researchers advocate the superiority of PSO over GA and highlight its capacity to solve complex problems thanks to its ease of implementation. In this paper, different versions of PSOs and GAs including our specific GA scheme are compared for FIR filter design. PSO generally outperforms standard GAs in some performance criteria, but our adaptive genetic algorithm is shown to be better on all criteria except CPU runtime. The study also underlines the importance of introducing intelligence in metaheuristics to make them more efficient by embedding self-tuning strategies. Furthermore, it establishes the potential complementarity of the approaches when solving this optimization problem.     

基于变换域全相位FIR自适应滤波算法

基于一种全相位FIR自适应滤波器,将重叠滤波思想引入变换域LMS算法,提出了DFT、DCT和DST变换域的带窗重叠自适应滤波算法(WO-TLMS).与传统的变换域LMS(TLMS)算法相比,WO-TLMS算法提高了收敛速度同时具有较低的稳态均方误差.理论分析了算法的收敛性,实验中通过和TLMS算法的比较验证了WO-TLMS算法的优越性.     

New algorithm for fir filter design with discrete coefficients

This paper develops a new algorithm based on the Projected Gradient Algorithm (PGA) for the design of FIR digital filters with “sum of power of two” coefficients. It is shown that the integer programming involved in the FIR filter design can be solved by this algorithm.It is compared with the reported method for a SemiDefinite Programming (SDP) relaxationbased design. The simulations demonstrate that the new algorithm often yields the similar error performances of the FIR filter design, but the average CPU time of this approach is significantly reduced.     

自适应加权FIR-Myriad混合滤波算法

针对FIR滤波器滤除脉冲噪声以及加权Myriad滤波器滤除高斯噪声的不足,提出基于FIR滤波器和加权WMy滤波器有效组合的一类新的非线性滤波器FIR-WMyH滤波器.利用神经网络中的反向传播算法,在均方误差准则下,推导了一个基于统计梯度的自适应算法.基于稳定α分布脉冲噪声模型下的仿真结果说明了该算法的良好的性能.     

采用SRM准则对RLS-FIR滤波器的改进

针对自适应FIR滤波器在非平稳条件下跟踪性能差的问题,提出采用结构风险最小化(SRM)准则来改进FIR滤波器的方法,并进行了分析和仿真实验.仿真结果表明:在平稳条件下,采用SRM准则的FIR滤波器收敛于Wiener最优解;在非平稳条件下,采用SRM准则的FIR滤波器跟踪性能大幅提高,优于自适应递推最小二乘(RIS-FI...     

Design and performance of adaptive systems based on structured stochastic optimization strategies

The theory and design of linear adaptive filters based on FIR filter structures is well developed and widely applied in practice. However, the same is not true for more general classes of adaptive systems such as linear infinite impulse response adaptive filters (MR) and nonlinear adaptive systems. This situation results because both linear IIR structures and nonlinear structures tend to produce multi-modal error surfaces for which stochastic gradient optimization strategies may fail to reach the global minimum. After briefly discussing the state of the art in linear adaptive filtering, the attention of this paper is turned to MR and nonlinear adaptive systems for potential use in echo cancellation, channel equalization, acoustic channel modeling, nonlinear prediction, and nonlinear system identification. Structured stochastic optimization algorithms that are effective on multimodal error surfaces are then introduced, with particular attention to the particle swarm optimization (PSO) technique. The PSO algorithm is demonstrated on some representative IIR and nonlinear filter structures, and both performance and computational complexity are analyzed for these types of nonlinear systems.     

Adaptive Laguerre-lattice filters

Adaptive Laguerre-based filters provide an attractive alternative to adaptive FIR filters in the sense that they require fewer parameters to model a linear time-invariant system with a long impulse response. We present an adaptive Laguerre-lattice structure that combines the desirable features of the Laguerre structure (i.e., guaranteed stability, unique global minimum, and small number of parameters M for a prescribed level of modeling error) with the numerical robustness and low computational complexity of adaptive FIR lattice structures. The proposed configuration is based on an extension to the IIR case of the FIR lattice filter; it is a cascade of identical sections but with a single-pole all-pass filter replacing the delay element used in the conventional (FIR) lattice filter. We utilize this structure to obtain computationally efficient adaptive algorithms (O(M) computations per time instant). Our adaptive Laguerre-lattice filter is an extension of the gradient adaptive lattice (GAL) technique, and it demonstrates the same desirable properties, namely, (1) excellent steady-state behavior, (2) relatively fast initial convergence (comparable with that of an RLS algorithm for Laguerre structure), and good numerical stability. Simulation results indicate that for systems with poles close to the unit circle, where an (adaptive) FIR model of very high order would be required to meet a prescribed modeling error, an adaptive Laguerre-lattice model of relatively low order achieves the prescribed bound after just a few updates of the recursions in the adaptive algorithm     

An iterative algorithm for optimal design of non-frequency-selective FIR digital filters

This paper proposes a novel iterative algorithm for optimal design of non-frequency-selective Finite Impulse Response （FIR） digital filters based on the windowing method. Different from the traditional optimization concept of adjusting the window or the filter order in the windowing design of an FIR digital filter, the key idea of the algorithm is minimizing the approximation error by successively modifying the design result through an iterative procedure under the condition of a fixed window length. In the iterative procedure, the known deviation of the designed frequency response in each iteration from the ideal frequency response is used as a reference for the next iteration. Because the approximation error can be specified variably, the algorithm is applicable for the design of FIR digital filters with different technical requirements in the frequency domain. A design example is employed to illustrate the efficiency of the algorithm.     

Oversampled sigma-delta LMS adaptive FIR filters

Two filter designs for adaptive least mean squares (LMS) filtering with sigma-delta modulated input signals are described. One implementation is multibit multiplier-free and operates entirely at the oversampling frequency of the sigma-delta signals, in the other design only the FIR filter operates at the oversampled frequency while the adaptive filtering algorithm is performed at the Nyquist rate. To circumvent any aliasing problems that may be caused by the downsampling process in the architecture and ensure convergence of the adaptive FIR filter. It is necessary to attenuate the high-frequency sigma-delta quantisation noise that is present. To perform this task a multiplier-free, multistage IIR filter structure is used that requires considerably fewer computations than an equivalent FIR filter. The two adaptive LMS filter designs are analysed and their performance is compared with a conventional PCM system in terms of achievable minimum MSE and adaptation speed