基于LMS算法的自适应滤波器仿真实现

为了达到最佳的滤波效果,使自适应滤波器在工作环境变化时自动调节其单位脉冲响应特性,提出了一种自适应算法:最小均方算法(LMS算法)。这种算法实现简单且对信号统计特性变化具有稳健性,所以获得了极为广泛的应用。针对用硬件实现LMS算法的自适应滤波器存在的诸多缺点,采用Matlab工具对基于LMS算法的自适应滤波器进行了仿真试验。仿真结果表明,应用LMS算法的自适应滤波器不仅可以实现对信号噪声的自适应滤除,还能用于系统识别。     

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.     

用SystemView仿真LMS自适应算法

最小均方误差(LMS)算法是自适应信号处理中最常用的算法.本文在给出LMS算法的原理的基础上,设计了一种单一频率的自适应陷波器的仿真方案.采用SystemView通信系统仿真工具,仿真了该自适应陷波器工作过程,给出了各点工作波形,并通过实验给出了不同参数条件下的陷波器收敛性能.实验结果表明,在合适的参数条件下,LMS算法可以兼顾收敛速度和稳态误差两方面的性能,实现性能良好的陷波器.同时,由于采用迭代算法,LMS算法更适合DSP或FPGA的数字实现.     

LFM信号的分数阶傅里叶域自适应滤波算法研究

对于线性调频信号(LFM)的滤波,采用处理平稳信号的方法对其滤波往往得不到很好的效果。本文利用了线性调频信号在分数傅里叶变换域上具有很好的时频聚焦性的特点,来实现信号在分数阶傅里叶域的自适应滤波,自适应滤波算法采用改进的步长LMS方法,对传统的LMS算法做出了改进,算法中步长处理中引入了一个限制因子,可以较好地解决算法收敛速度和稳态失调量之间的矛盾。仿真结果表明,此算法在处理分数阶域的LFM信号滤波比传统的LMS算法有较好的滤波效果。      

一种优化的激光陀螺自适应抖动剥除算法

去除激光陀螺输出信号中的抖动信号是机械抖动偏频激光陀螺(MDRLG)信号处理电路的主要工作之一。文章对LMS算法中误差因子响应特性对LMS自适应抖动剥除效果的影响进行了研究。为改善抖动剥除效果,在自适应抖动剥除算法中引入了带通滤波器,并采用基于FPGA的激光陀螺自适应抖动剥除系统对误差因子采用带通滤波时的静态性能和动态响应特性进行了测试。试验结果表明,误差因子采用带通滤波时,LMS自适应抖动剥除的收敛速度和收敛精度都要优于差分运算,并且二者具有相同的动态频率响应曲线。     

带有噪声相关性的自适应噪声抵消系统仿真研究

本文为了在语音信号处理中能消除含噪语音信号中的背景噪音,采用自适应信号处理的理论和技术来达到提高语音信号质量的目的。通过介绍自适应滤波器原理,在对自适应滤波器相关理论研究的基础上,研究了LMS自适应滤波算法,并对LMS自适应算法进行了分析。同时为了使输入的参考信号与噪声相关,加入分离周期信号与带有窄带干扰抑制的宽带信号。通过分析仿真结果表明基于LMS算法的自适应噪声抵消技术可以有效地抵消正弦干扰信号,同时加入宽带信号中的周期性噪声,在没有另外的与噪声相关的参考信号的情况下,可以使用自适应噪声抵消系统来消除这种同期性干扰噪声。     

Analytical model for the first and second moments of an adaptiveinterpolated FIR filter using the constrained filtered-X LMS algorithm

The authors present an analytical model for the mean weight behaviour and weight covariance matrix of an adaptive interpolated FIR filter using the LMS algorithm to adapt the filter weights. The particular structure of this adaptive filter determines that special analytical considerations must be used. First, the introduction of an interpolating block cascaded with the adaptive sparse filter requires that the input signal correlations must be considered. It is well known that such correlations are disregarded by the independence theory, which is the basis for the analysis of the LMS algorithm adapting FIR structures. Secondly a constrained analysis is used to deal mathematically with the sparse nature of the adaptive section. Experimental results demonstrate the effectiveness of the proposed analytical models as compared with the results obtained by classical analysis     

基于最小均方自适应滤波器的无线光通信接收性能分析

无线光通信(OWC)系统采用大气通道作为传输媒介,而大气湍流效应引入了与信号强度有关的乘性噪声。为了消除乘性噪声所引起的信号衰落,分析并给出了基于最小均方(LMS)自适应滤波器的判决门限更新算法和稳态的抽头权系数相关矩阵算法。通过理论分析和计算机仿真,讨论了最小均方滤波器及其参数对无线光通信接收性能的影响。结果表明,采用非因果滤波器的无线光通信系统对湍流噪声具有明显的抑制作用。在弱湍流情况下,基于自适应最小均方滤波器的系统误码率(BER)低于10-8,可以满足网络通信的要求。通过分析不同滤波器阶数对误码率的影响表明,所采用的255阶的非因果结构的最小均方滤波器是最优的结构。     

Adaptive filter for event-related bioelectric signals using animpulse correlated reference input: comparison with signal averagingtechniques

Many bioelectric signals result from the electrical response of physiological systems to an impulse that can be internal (ECG signals) or external (evoked potentials). In this paper an adaptive impulse correlated filter (AICF) for event-related signals that are time-locked to a stimulus is presented. This filter estimates the deterministic component of the signal and removes the noise uncorrelated with the stimulus, even if this noise is colored, as in the case of evoked potentials. The filter needs two inputs: the signal (primary input) and an impulse correlated with the deterministic component (reference input). We use the LMS algorithm to adjust the weights in the adaptive process. First, we show that the AICF is equivalent to exponentially weighted averaging (EWA) when using the LMS algorithm. A quantitative analysis of the signal-to-noise ratio improvement, convergence, and misadjustment error is presented. A comparison of the AICF with ensemble averaging (EA) and moving window averaging (MWA) techniques is also presented. The adaptive filter is applied to real high-resolution ECG signals and time-varying somatosensory evoked potentials.     

ULV and generalized ULV subspace tracking adaptive algorithms

Traditional adaptive filters assume that the effective rank of the input signal is the same as the input covariance matrix or the filter length N. Therefore, if the input signal lives in a subspace of dimension less than N, these filters fail to perform satisfactorily. In this paper, we present two new algorithms for adapting only in the dominant signal subspace. The first of these is a low-rank recursive-least-squares (RLS) algorithm that uses a ULV decomposition (Stewart 1992) to track and adapt in the signal subspace. The second adaptive algorithm is a subspace tracking least-mean-squares (LMS) algorithm that uses a generalized ULV (GULV) decomposition, developed in this paper, to track and adapt in subspaces corresponding to several well-conditioned singular value clusters. The algorithm also has an improved convergence speed compared with that of the LMS algorithm. Bounds on the quality of subspaces isolated using the GULV decomposition are derived, and the performance of the adaptive algorithms are analyzed     

LMS自适应滤波器干扰方法

自适应滤波器能有效地提高雷达在复杂电磁环境下的适应能力,在雷达信号处理机中得到广泛的运用,其核心是使用自适应算法,将滤波器设计成根据目标对照射信号的响应,及外界的电磁环境的变化等因素,调节滤波器的自身结构参数,最终趋于稳态的维纳滤波器,实现对目标的最优检测。文中主要分析LMS自适应算法中存在的步长固定等缺点,设计干扰信号,并用计算机仿真验证其对LMS滤波器的干扰效果,该干扰信号可使自适应滤波器无法实现结构上的自动优化,降低其雷达的工作效能。     

Analysis of stochastic gradient tracking of time-varying polynomialWiener systems

This paper presents analytical and Monte Carlo results for a stochastic gradient adaptive scheme that tracks a time-varying polynomial Wiener (1958) system [i.e., a linear time-invariant (LTI) filter with memory followed by a time-varying memoryless polynomial nonlinearity]. The adaptive scheme consists of two phases: (1) estimation of the LTI memory using the LMS algorithm and (2) tracking the time-varying polynomial-type nonlinearity using a second coupled gradient search for the polynomial coefficients. The time-varying polynomial nonlinearity causes a time-varying scaling for the optimum Wiener filter for Phase 1. These time variations are removed for Phase 2 using a novel coupling scheme to Phase 1. The analysis for Gaussian data includes recursions for the mean behavior of the LMS algorithm for estimating and tracking the optimum Wiener filter for Phase 1 for several different time-varying polynomial nonlinearities and recursions for the mean behavior of the stochastic gradient algorithm for Phase 2. The polynomial coefficients are shown to be accurately tracked. Monte Carlo simulations confirm the theoretical predictions and support the underlying statistical assumptions     

Convergence analysis of adaptive filtering algorithms with singulardata covariance matrix

The paper provides a rigorous analysis of the behavior of adaptive filtering algorithms when the covariance matrix of the filter input is singular. The analysis is done in the context of adaptive plant identification. The considered algorithms are LMS, RLS, sign (SA), and signed regressor (SRA) algorithms. Both the signal and weight behavior of the algorithms are considered. The signal behavior is evaluated in terms of the moments of the excess output error of the filter. The weight behavior is evaluated in terms of the moments of the filter weight misalignment vector. It is found that the RLS and SRA diverge when the input covariance matrix is singular. The steady-state signal behavior of the LMS and SA can be made arbitrarily fine by using sufficiently small step sizes of the algorithms. Indeed, the long-term average of the mean square excess error of the LMS is proportional to the algorithm step size. The long-term average of the mean absolute excess error of the SA is proportional to the square root of the algorithm step size. On the other hand, the steady-state weight behavior of both the LMS and SA have biases that depend on the weight initialization. The analytical results of the paper are supported by simulations     

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     

LMS和归一化LMS算法收敛门限与步长的确定

从LMS算法失调量的准确表达式出发,根据输入信号特征值分布重新研究了LMS,归一化LMS(Normalized LMS,NLMS)算法收敛的必要条件,推导出LMS和NLMS 算法收敛的步长门限,并分析了输入信号特征值分布、滤波器阶数对算法收敛步长门限的影响,推导出满足性能失调下步长的自适应计算公式,减小了应用 LMS,NLMS算法时步长选取的盲目性,与已有的算法相比,具有计算简单、实用、自适应性能强,同时可获得满意失调量的特点,计算机模拟结果表明该方法的正确性。     

语音信号增强系统设计与仿真

在信号处理中语音增强是一个重要分支,针对语音信号不得不在噪声和干扰环境下通信的现状,采用自适应滤波算法设计了语音信号噪声和干扰抑制系统,首先对LMS算法进行了推导,并且对噪声和干扰环境下的自适应滤波器性能进行了仿真分析,仿真结果表明:该设计抑制干扰和噪声的性能较好,对语音信号的增强明显,为该系统在硬件上实现提供了理论基础。     

A class of order statistic LMS algorithms

Conventional gradient-based adaptive filters, as typified by the well-known LMS algorithm, use an instantaneous estimate of the error-surface gradient to update the filter coefficients. Such a strategy leaves the algorithm extremely vulnerable to impulsive interference. A class of adaptive algorithms employing order statistic filtering of the sampled gradient estimates is presented. These algorithms, dubbed order statistic least mean squares (OSLMS), are designed to facilitate adaptive filter performance close to the least squares optimum across a wide range of input environments from Gaussian to highly impulsive. Three specific OSLMS filters are defined: the median LMS, the average LMS, and the trimmed-mean LMS. The properties of these algorithms are investigated and the potential for improvement demonstrated. Finally, a general adaptive OSLMS scheme in which the nature of the order-statistic operator is also adapted in response to the statistics of the input signal is presented. It is shown that this can facilitate performance gains over a wide range of input data types     

Analysis of Convergence of a Frequency-Domain LMS Adaptive Filter Implemented as a Multi-Stage Adaptive Filter

We present an analysis of the convergence of the frequency-domain LMS adaptive filter when the DFT is computed using the LMS steepest descent algorithm. In this case, the frequency-domain adaptive filter is implemented with a cascade of two sections, each updated using the LMS algorithm. The structure requires less computations compared to using the FFT and is modular suitable for VLSI implementations. Since the structure contains two adaptive algorithms updating in parallel, an analysis of the overall system convergence needs to consider the effect of the two adaptive algorithms on each other, in addition to their individual convergence. Analysis was based on the expected mean-square coefficient error for each of the two LMS adaptive algorithms, with some simplifying approximations for the second algorithm, to describe the convergence behavior of the overall system. Simulations were used to verify the results.     

一种用于QAM解调信号的LMS自适应均衡器

设计了一种用于QAM(Quadrature Amplitude Modulation)解调信号的LMS自适应均衡器。此均衡器采用线性自适应算法中的最小均方算法(LMS)．其结构由线性横向滤波器和需要训练序列的滤波器抽头系数更新模块组成．它可实现16／64／256点的QAM解调。利用MATLAB／Simulink对LMS自适应均衡器的收敛速度、误码率等指标进行仿真模拟,仿真结果表明,此LMS自适应均衡器对通过非理想信道的QAM传输信号具有较好的均衡性能。     

平稳环境下LMS算法优化分析及其FPGA实现

高速自适应滤波器的设计及其现场可编程门阵列（FPGA）实现一直是信号处理领域研究的热点。普通最小均方误差算法（LMS）自适应滤波器运行速度受到计算滤波输出和系数更新时间限制,制约了其信号处理的速度。对于平稳环境,通过对自适应滤波器系数更新方程进行前瞻和松弛近似,对LMS自适应算法进行了优化分析,得到了一种改进的LMS自适应滤波器结构。利用DSP Builder工具建立了四阶改进结构的LMS自适应滤波器模型并进行了一系列的仿真,结合多种电子设计自动化（EDA）工具,最终在EP2C35型FPGA上得到了最高响应速度60.07 MHz的高速自适应滤波器。结果表明,改进的LMS自适应滤波器速度较一般结构滤波器快,但耗费了较多逻辑资源。