一种基于误差相关函数的改进变步长LMS算法

最小均方算法的收敛速度和稳态误差之间存在矛盾,为此人们提出了各种变步长LMS算法,其中E-LMS算法是将步长与瞬时误差平方相关联,R-LMS算法是将步长与误差的相关函数相关联。E-LMS算法的抗噪性能较差,在低信噪比条件下性能明显变差,R-LMS算法对突变系统的跟踪能力较差。为此文中给出了一种改进的,基于误差相关函数的VSS-LMS算法,该方法利用E-LMS算法的控制步长策略提高算法的跟踪能力。计算机仿真结果显示,该算法能够同时满足抗噪和跟踪两种要求。     

基于频带间相关性的加权噪声功率谱估计

噪声功率谱估计是语音增强系统的一个重要组成部分。本文在加权噪声估计的基础上,考虑了带噪语音在相邻频带间的相关性,提出了一种新的噪声功率谱估计算法。该算法保留了加权噪声估计算法的优点,利用频域平滑及时域平滑后的带噪语音来求加权因子,能够更好地区分弱语音与噪声,尤其是对强语音后的弱语音与噪声区分更明显,从而具有更快的跟踪速度及更少的噪声过估计。客观实验和主观实验都证实了本文提出的算法的有效性。     

Comparative tracking performance of the LMS and RLS algorithms forchirped narrowband signal recovery

This paper studies the comparative tracking performance of the recursive least squares (RLS) and least mean square (LMS) algorithms for time-varying inputs, specifically for linearly chirped narrowband input signals in additive white Gaussian noise. It is shown that the structural differences in the implementation of the LMS and RLS weight updates produce regions where the LMS performance exceeds that of the RLS and other regions where the converse occurs. These regions are shown to be a function of the signal bandwidth and signal-to-noise ratio (SNR). LMS is shown to place a notch in the signal band of the mean lag filter, thus reducing the lag error and improving the tracking performance. For the chirped signal, it is shown that this produces smaller tracking error for small SNR. For high SNR, there is a region of signal bandwidth for which RLS will provide lower error than LMS, but even for these high SNR inputs, LMS always provides superior performance for very narrowband signals     

一种用于自适应噪声抵消的变步长LMS算法

为了提高LMS自适应滤波算法的性能,在分析已有变步长算法的基础上进行了一些改 进。改进算法用误差信号的自相关来调节步长以实现对不相关噪声的更好抑制,且采 用先固定后变化的方法控制步长,兼顾了暂态和稳态性能。利用改进算法进行了自适应噪声 抵消的仿真实验,结果表明,基于改进变步长LMS算法的自适应噪声抵消器 能有效抵制噪声干扰,对含噪信号具有良好的消噪能力。     

一种约束稳定性最小均方噪声对消算法

研究了最小均方误差(LMS)算法、归一化的最小均方(NLMS)算法及变步长NLMS算法在自适应噪声干扰抵消器中的应用,针对目前这些算法在噪声对消器应用中的缺点,将约束稳定性最小均方(CS-LMS)算法应用到噪声处理中,并进一步结合变步长的思想提出来一种新的变步长CS-LMS算法。通过MATLAB进行仿真分析,结果证实提出的算法与其他算法相比,能很好地滤除掉噪声从而得到期望信号,明显的降低了稳态误差,并拥有好的收敛速度。     

Generalized Robust Multichannel Frequency‐Domain LMS Algorithms for Blind Channel Identification

Recently, several noise‐robust adaptive multichannel LMS algorithms have been proposed based on the spectral flatness of the estimated channel coefficients in the presence of additive noise. In this work, we propose a general form for the algorithms that integrates the existing algorithms into a common framework. Computer simulation results are presented and demonstrate that a new proposed algorithm gives better performance compared to existing algorithms in noisy environments.     

A set of algorithms linking NLMS and block RLS algorithms

This paper describes a set of block processing algorithms which contains as extremal cases the normalized least mean squares (NLMS) and the block recursive least squares (BRLS) algorithms. All these algorithms use small block lengths, thus allowing easy implementation and small input-output delay. It is shown that these algorithms require a lower number of arithmetic operations than the classical least mean squares (LMS) algorithm, while converging much faster. A precise evaluation of the arithmetic complexity is provided, and the adaptive behavior of the algorithm is analyzed. Simulations illustrate that the tracking characteristics of the new algorithm are also improved compared to those of the NLMS algorithm. The conclusions of the theoretical analysis are checked by simulations, illustrating that, even in the case where noise is added to the reference signal, the proposed algorithm allows altogether a faster convergence and a lower residual error than the NLMS algorithm. Finally, a sample-by-sample version of this algorithm is outlined, which is the link between the NLMS and recursive least squares (RLS) algorithms     

Block adaptive filters with deterministic reference inputs forevent-related signals: BLMS and BRLS

Adaptive estimation of the linear coefficient vector in truncated expansions is considered for the purpose of modeling noisy, recurrent signals. Two different criteria are studied for block-wise processing of the signal: the mean square error (MSE) and the least squares (LS) error. The block LMS (BLMS) algorithm, being the solution of the steepest descent strategy for minimizing the MSE, is shown to be steady-state unbiased and with a lower variance than the LMS algorithm. It is demonstrated that BLMS is equivalent to an exponential averager in the subspace spanned by the truncated set of basis functions. The block recursive least squares (BRLS) solution is shown to be equivalent to the BLMS algorithm with a decreasing step size. The BRLS is unbiased at any occurrence number of the signal and has the same steady-state variance as the BLMS but with a lower variance at the transient stage. The estimation methods can be interpreted in terms of linear, time-variant filtering. The performance of the methods is studied on an ECG signal, and the results show that the performance of the block algorithms is superior to that of the LMS algorithm. In addition, measurements with clinical interest are found to be more robustly estimated in noisy signals     

An adaptive estimation of periodic signals using a Fourier linearcombiner

Presents an adaptive algorithm for estimating from noisy observations, periodic signals of known period subject to transient disturbances. The estimator is based on the LMS algorithm and works by tracking the Fourier coefficients of the data. The estimator is analyzed for convergence, noise misadjustment and lag misadjustment for signals with both time invariant and time variant parameters. The analysis is greatly facilitated by a change of variable that results in a time invariant difference equation. At sufficiently small values of the LMS step size, the system is shown to exhibit decoupling with each Fourier component converging independently and uniformly. Detection of rapid transients in data with low signal to noise ratio can be improved by using larger step sizes for more prominent components of the estimated signal. An application of the Fourier estimator to estimation of brain evoked responses is included     

一种改进的变步长LMS自适应算法

传统的最小均方误差(LMS)算法难以同时获取较快的收敛速度和较小的稳态误差,而变步长LMS算法可获得二者之间的平衡。对已有的一些变步长LMS算法进行了分析,在变系数步长(VFSS)算法的基础上,引入输入信号因子,并建立步长因子与误差信号之间新的非线性函数关系,提出一种改进的变步长LMS算法,该算法不仅继承了VFSS算法在低信噪比环境下抗噪声性能好的特点,而且能够快速跟踪系统的变化,仿真结果表明改进算法的性能优于现有算法。     

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。     

Vision-based road-traffic monitoring sensor

Current techniques for road-traffic monitoring rely on sensors that have limited capabilities and are often both costly and disruptive to install. The use of video cameras (many of which are already installed to survey road networks), coupled with computer vision techniques, offers an attractive alternative to current sensors. Vision-based sensors have the potential to measure a greater variety of traffic parameters (e.g. entry/exit statistic, journey times and incident detection) while installation and maintenance may be performed without disruption to traffic flow. Work on a model based approach for locating vehicles in images of complex road scenes is presented. The location of the vehicle in the image is transformed to the vehicle's position and orientation in the real world while the deformable vehicle model allows the vehicle's principal dimensions to be measured. This data may be passed to a high level tracking algorithm to extract traffic parameters such as vehicle speed, vehicle count, and junction entry/exit statistics. The principal dimensions may be used to classify the vehicle within categories such as car, van or bus. The system could also be used as a boot-strap process for faster, but perhaps less robust, tracking algorithms. The key features of the system are described and results from testing it on images from real traffic scenes are presented     

自适应滤波器的修正的时域正交性算法及其应用

本文绘出了一个自适应滤波器的修正的时域正交性算法.计算机模拟结果表明,该算法有两个特性:一、相同的收敛因子c可以在比较宽范围的输入信号信噪比内使用。二、应用这种方法的噪声消除器的一个实用特性是,对于相同的期待信号,输入的参考信号的功率变化,不影响收敛因子c值的选择,这就使这种噪声消除器的应用更灵活。文中给出了这种算法用于谱估值时的特性,并与LMS算法进行了比较。     

一种新的变步长LMS算法

在对基本LMS算法分析的基础上,通过构造步长因子μ与误差信号e（n）之间的非线性函数,提出一种新的变步长最小均方误差（LMS）算法,并且分析了参数的取值对算法性能的影响。该算法通过调整步长参数,使权向量达到最优,有效改善了收敛速度与稳态误差的性能。理论分析和仿真结果表明,与基本LMS算法以及部分同类变步长LMS算法相比,该算法具有更快的收敛速度和更小的稳态误差,进一步验证了新算法优于这里所述其他算法。     

Sinusoidal interference rejection analysis of an LMS adaptivefeedforward controller with a noisy periodic reference

This paper applies a new vector subspace model to determine the non-Wiener solutions of the LMS algorithm when the reference input is an arbitrary noisy periodic signal. The LMS weights are modeled as a deterministic time-varying mean plus a zero-mean fluctuating part. For the mean weight, each harmonic component of the periodic reference is shown to excite only a two-dimensional (2-D) subspace of the N-dimensional tap weight space. The fluctuating part of the weight is due to the reference noise that excites the full N-dimensional space. The power spectrum of the error is computed using the correlation function of the weight fluctuations. The error is shown to be primarily the sum of a white noise and the canceled periodic residual for an independent noise vector model. When the effects of the tapped delay line are incorporated in the model, the noise becomes colored with a spectrum centered about the canceled sinusoidal frequency. This weight model significantly simplifies the understanding of the non-Wiener behavior and can be applied to active noise cancellation problems when filters appear in the cancellation loop     

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。     

Adaptive recovery of a chirped sinusoid in noise. II. Performanceof the LMS algorithm

For pt.I see ibid., vol.39, no. 3, p.583-94 (1991). The authors present a methodology for evaluating the tracking behavior of the least-mean square (LMS) algorithm for the nontrivial case of recovering a chirped sinusoid in additive noise. A complete closed-form analysis of the LMS tracking properties for a nonstationary inverse system modeling problem is also presented. The mean-square error (MSE) performance of the LMS algorithm is calculated as a function of the various system parameters. The misadjustment or residual of the adaptive filter output is the excess MSE as compared to the optimal filter for the problem. It is caused by three errors in the adaptive weight vector: the mean lag error between the (time-varying mean) weight and the time-varying optimal weight; the fluctuations of the lag error; and the noise misadjustment which is due to the output noise. These results are important because they represent a precise analysis of a nonstationary deterministic inverse modeling system problem with the input being a colored signal. The results are in agreement with the form of the upper bounds for the misadjustment provided by E. Eweda and O. Macchi (1985) for the deterministic nonstationarity     

迭代变步长LMS算法及性能分析

针对固定步长LMS(Least Mean Square)算法(FXSSLMS)不能同时满足快速收敛和小稳态失调误差的问题,该文提出了迭代变步长LMS算法(IVSSLMS)。与已有的变步长LMS算法(VSSLMS)不同,该算法的步长因子不再是由输出误差信号控制,而是建立了与迭代时间的改进Logistic函数非线性关系,克服了定步长算法收敛慢及已有变步长算法抗噪声干扰能力差的问题。最后从理论上分析了算法的性能,给出了其参数取值方法。理论分析和仿真均表明,所提算法能够在快速收敛情况下获得小的稳态失调误差,在有色噪声干扰下稳态失调误差比已有算法降低了约7 dB。     

Constrained LMS adaptive algorithm

A new least-mean-squares (LMS) adaptive algorithm is developed in the letter. This new algorithm solves a specific variance problem that occurs in LMS algorithms in the presence of high noise levels and when the input signal is bandlimited. Quantitative results in terms of an accuracy measure of a finite impulse response (FIR) system identification are presented.     

一种强跟踪抗噪声自适应滤波算法

通过分析几种典型的LMS算法,结合几种典型算法的优点,在不同时期采用不同的控制步长策略,本文提出基于相关箕舌线的自适应滤波算法(CTCLMS算法),并对其性能指标进行了比较仿真。在同一仿真条件下的仿真结果表明:CTCLMS算法除了具有收敛速度快,稳态误差小,计算简单的特点外,还同时具有了强跟踪能力和强抗噪声能力,解决了系统跟踪能力和抗噪声性能之间的矛盾。