基于LMS算法的自适应组合滤波器中英文翻译
自适应窗口高斯滤波法
自适应窗口高斯滤波法(中英文版)英文文档:Adaptive Window Gaussian Filter MethodThe adaptive window Gaussian filter method is a technique used in image processing to smooth or blur an image while preserving its edges.This method is particularly useful in situations where the size of the image or the level of noise is unknown, as it adjusts the window size automatically to achieve the best filtering results.The main idea behind the adaptive window Gaussian filter is to vary the size of the window used for filtering based on the local characteristics of the image.In areas with high edge intensity, a smaller window is used to better preserve the edges, while in areas with lower edge intensity, a larger window is used to effectively reduce noise.The adaptive window Gaussian filter method can be implemented using various algorithms, such as the median filter, which is a non-linear digital filtering technique that is often used to remove noise from an image or signal.The median filter works by replacing each pixel"s value with the median value of the pixels in the window.This helps to preserve the edges while reducing the overall noise level.Another popular algorithm used in the adaptive window Gaussian filter method is the双边滤波器(bilateral filter).The bilateral filter is a non-linear filter that takes into account both the spatial closeness and the intensity similarity of the pixels in the window.This helps to preserve the edges while also reducing noise, making it a popular choice for image denoising and smoothing.In conclusion, the adaptive window Gaussian filter method is a powerful technique for image processing that automatically adjusts the window size to achieve the best filtering results.It can be implemented using various algorithms, such as the median filter and the bilateral filter, and is particularly useful in situations where the size of the image or the level of noise is unknown.中文文档:自适应窗口高斯滤波法自适应窗口高斯滤波法是一种在图像处理中用于平滑或模糊图像的同时保留边缘的技术。
基于LMS算法的自适应滤波器设计
基于LMS算法的自适应滤波器设计自适应滤波器是信号处理中常用的一种技术,可以根据输入信号的统计特性来调整滤波器参数,以实现信号的去噪、谱线增强等功能。
LMS (Least Mean Square,最小均方误差)算法是自适应滤波器中最常用的一种算法,它通过调整滤波器的权值,使得滤波器的输出信号与期望输出信号之间的均方误差最小。
本文将详细介绍基于LMS算法的自适应滤波器设计。
首先,我们先来了解LMS算法的原理。
LMS算法的核心思想是通过不断迭代调整滤波器的权值,使得滤波器的输出信号最小化与期望输出信号之间的均方误差。
算法的迭代过程如下:1.初始化滤波器权值向量w(0)为0;2.对于每个输入信号样本x(n),计算滤波器的输出信号y(n);3.计算实际输出信号y(n)与期望输出信号d(n)之间的误差e(n);4.根据误差信号e(n)和输入信号x(n)来更新滤波器的权值向量w(n+1);5.重复步骤2-4,直到满足停止条件。
在LMS算法中,滤波器的权值更新公式为:w(n+1)=w(n)+μ*e(n)*x(n)其中,w(n+1)为更新后的权值向量,w(n)为当前的权值向量,μ为步长参数(控制权值的调整速度),e(n)为误差信号,x(n)为输入信号。
1.确定输入信号和期望输出信号的样本数量,以及步长参数μ的值;2.初始化滤波器的权值向量w(0)为0;3.依次处理输入信号样本,在每个样本上计算滤波器的输出信号y(n),并计算出误差信号e(n);4.根据误差信号e(n)和输入信号x(n)来更新滤波器的权值向量w(n+1);5.重复步骤3-4,直到处理完所有的输入信号样本;6.得到最终的滤波器权值向量w,即为自适应滤波器的设计结果。
在实际应用中,自适应滤波器设计的性能往往与步长参数μ的选择密切相关。
较小的步长参数会使得权值更新速度过慢,容易出现收敛慢的问题;而较大的步长参数可能导致权值在稳定后开始震荡,使得滤波器的性能下降。
基于LMS自适应滤波器的设计
基于LMS 自适应滤波器的设计姜泉璐,汪立新,吕永佳,吴玉彬,叶军(第二炮兵工程学院陕西西安710025)摘要:针对在惯组时间序列的插值和预测分析中,不可避免地会引入一些误差,致使预报结果不能反映惯组实际状况的缺点,基于自适应滤波能够提供卓越的滤波性能和自动地调节现时刻滤波器参数的特点,设计了一种基于LMS 自适应滤波器,以适应信号和噪声未知的或随时间变化的统计特性,从而实现最优滤波。
仿真结果表明该方法是有效的。
关键词:LMS ;自适应;滤波器;最优滤波;有效性中图分类号:TJ765文献标识码:A文章编号:1674-6236(2011)14-0067-03Design of adaptive filter based on LMSJIANG Quan -lu ,WANG Li -xin ,LV Yong -jia ,WU Yu -bin ,YE Jun(The Second Artillery Engineering College ,Xi ’an 710025,China )Abstract:For the drawbacks that in the analysis of the Inertial Measurement Unit (IMU )time series interpolation andprediction ,it will inevitably introduce some errors and make the forecast results cannot reflect the actual condition of the IMU.Based on the characteristics that adaptive filter can provide excellent filtering performance and automatically adjust the filter current parameters ,an adaptive filter based on the LMS is designed ,to adapt to unknown signal and noise or time -varying statistical properties ,realizing the optimal filter.Simulation results show this method is effective.Key words:LMS ;adaptive ;filter ;optimal filtering ;effectiveness收稿日期:2011-04-23稿件编号:201104114作者简介:姜泉璐(1985—),男,山东曹县人,硕士研究生。
基于LMS算法的自适应滤波器的设计
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摘 要:设计了基于 LMS 算法的 FIR 自适应滤波器,采用 MATLAB 仿真软件对其进行 了仿真。根据自顶向下的设计流程,采用 DDS 技术和流水线设计方法,在 Quartus II 开 发平台上采用 Verilog HDL 语言对其进行了硬件实现。实验结果证明该设计比普通自 适应滤波器具有结构简单、易于实现、滤波速度快的优点.
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LMS 中英文对照表位置原文简体中文繁体中文菜单File 文件New 新建Open 打开Reopen 重新打开Save 保存SaveAs 另存为Revert 返回Load QuickSet File 载入快速设置文件Save QuickSet File夹保存快速设置文件Load GraphSetup File 载入图表设置文件Save GraphSetup File 保存图表设置文件Print 打印Editor 编辑器Preferences 参数选择Exit 退出Graph 图表Parameters 参数Curve Library 曲线库Notes & Comments 附注&注释Analyzer 分析仪Sweep Start/Stop 扫频开始/停止Osc On/Off Osc 开/关RLC meter RLC 表Microphone Setup 麦克风设置PAC Interface PAC接口Micro Run 宏运行Calibration 剪贴板Processing 处理Unary Math Operations 一元数学操作Binary Math Operations 二元数学操作Minimum Phase Transform 最小相位转换Delay Phase Transform 延迟相位转换Group Delay Transform 组延迟转换Inv Fast Fourier Transform 快速反傅立叶转换Fast Fourier Transform 快速傅立叶转换Speaker Parameters 喇叭参数Tail Correction 尾部修正Data Transfer 数据合并Data Splice 数据接合Data Realign 数据重组Curve Averaging 数据提升Curve Compare 数据对照Curve Integration 数据综合Utilities 实用程序Import Curve Data File 导入曲线数据文件Export Curve Data File 导出曲线数据文件Export Graphics to File 导出图表到文件Export Graphics to Clipboard 导出图表到剪贴板Curve Capture 曲线捕捉Curve Editor 曲线编辑器Macro Editor 宏编辑器MDF Editor MDF 编辑器Polar Convertor 极性转换器View Clipboard 查看剪贴板Transducer Model Derivation 传感器模式导出Scale 刻度Parameters 参数Auto 自动Up 向上Down 向下View 查看Zoom In 放大Zoom Out 缩小Zoom 缩放Redraw 刷新ToolBars 工具栏Show All 全部显示Hide All 全部隐藏ToolBox 工具框Help 帮助Contents 内容Index 索引Glossary 术语表About Modules 关于模块About Program 关于软件对话框(Graph Parameters)Frame 边框Background 背景Note Underline 注释下划线Large Frame Line 大边框线Small Frame Line 小边框线Grid 格子Border Line 边界线Major Div 主格Minor Div 次格Font 字体Title Block 工程明细表Map Legend 映射图例Note List 注释列表Graph Title 图表标题Scale Vertical 垂直刻度Scale Horizontal 水平刻度Typeface 字体Style 字形Size 大小Color 颜色(Curve Library)Data Curve 数据曲线Same Line Type 统一线型Left(Magnitude) 左坐标(量级)Right Lighter 右坐标浅色Right(Phase) 右坐标(相位)Info 信息Horz Data Range 频宽范围Left Vert 左坐标Right Vert 右坐标Points 点Style 类型Width 宽度Color 颜色(Note&Comments)Left Page 左页面Right Page 右页面Title Block Data 工程图明细表Person 个人Company 公司Automatic Curve Info Notes 自动曲线信息标注(Analyzer Parameters)Oscillator 振荡器Output Level 输出水平Frequency 频率Mode 扫频模式Hi Speed Data 高速数据Precision Data 精密数据Gating 门控Off 关On 开Meter 仪表Data 数据Value 值Source 来源Freq 频率Sweep 扫频Lo Freq 低频Hi Freq 高频Direction 方向Control 控制Pulse 脉冲Gate Time Calculator 门控计时器Meter Filter 滤波器Filter Function 滤波函数Track Ratio 音轨率Gate Timing 门控时间(RLC Meter)Measurement 测量Resistance 电阻Inductance 电感Capacitance 电容Impedance 阻抗Limit Testing 测试限制Enable 启用MinValue 最小值Max Value 最大值(Microphone Setup)Mic Input 麦克风输入Line Input 信号输入Acoustic Ref 声学参数Electric Ref 电学参数Serial 序列号Author 制造商Date 日期Load 载入(PAC Interface)Serial Port 串行端口Linking 正在链接Start Link 开始链接Automatic Link 自动链接Baud Rate 波特率System Power 系统供电Power Source 供电电源Battery Status 电池状态Charge Amps 功放充电System Status 系统状态Link Status 链接状态Port 端口Voltage 电压Battery V oltage 电池电压External V oltage 外部电压(Analyzer Calibration)Internal 内部External 外部Parameter Under Test 测试参数Results 结果(Unary Math Operations)Library Curve 曲线库Operation 操作Magnitude Offset 幅度补偿Phase Offset 相位偏移Delay Offset 延迟偏移Exponentiation 求幂Smooth Curve 平滑曲线Frequency Translation 频率转换Multiply by 乘Divide 除Real 正弦Imag(sin) 余弦(Binary Math Operations)Mul 乘Div 除Add 加Sub 减Operand 操作数(Minimum Phase Transform)Asymptotic Slope at Hi Freq Limit 频率上限斜率Asymptotic Slope at Lo Freq Limit 频率下限斜率Automatic Tail Correction/Mirroring for Impedance Curves 自动尾部修正/阻抗曲线镜像(Delay Phase Transform)Source Curve with Delay Data 来源延迟数据曲线Result Curve for Phase Data 结果相位数据曲线(Group Delay Transform)Source Curve with Phase Data 来源相位曲线Result Curve for Group Delay 结果组延迟曲线(Inverse Fast Fourier Transform)Linear Frequency Points 线性频率点Frequency Domain Data 频率范围数据Result Impulse Curve 结果推动曲线Time Domain Data 时间范围数据Result Step Curve 结果步骤曲线(Speaker Parameters)Method 方法Single Curve 单曲线Double Curve 双曲线Reference Curve 参考曲线Standard 标准Estimate 估算Optimize 优化Simulate 模拟Model Simulation 模拟模型Copy Binary to Clipboard 复制二元数据到剪贴板Copy Text to Clipboard 复制文本到剪贴板(Tail Correction)Slope Lo 低频(Data Splice)Source Curve for Higher Data 来源高数据曲线Source Curve for Lower Data 来源低数据曲线Horz Splice Transition 水平接合转换(Data Realign)Linear 线性Log 对数Horz Lo Limit 水平下限Horz Hi Limit 水平上限Interpolation 插补Quadratic 平方Cubic 立方(Curve Averaging)Scalar 标量Vector 矢量(Curve Compare)Test Parameters 测试参数Absolute 绝对Relative 相对Tolerance 公差Relative Flatness 相对平面(Curve Integration)Average 平均Integrate 叠加(Import Curve Data)Left Vert Data 左坐标数据Right Ver Data 右坐标数据Polar Freq 极性频率Units 单位Prefix 前缀Curve Entry 曲线条目Special Processing 特殊处理Skip First Data Column 跳过首个数据列(Export Graphics)Artwork 作品Graph Name 图表名称Format 格式Raster 光栅Image Pixel Width 图像宽度(像素) Image Pixel Height 图像高度(像素) Image Bits per Pixels 图像位每像素Image Bytes 图像字节数(Clipboard Graphics Transfer)Enhanced Metafile 增强元文件Bitmap 位图(Curve Capture)Polar Freq 极性频率Graph Image 图表图像Reference Point Data 参考点数据Upper Right 右上Scan Direction 扫描方向Top to Bottom 顶到底Bottom to Top 底到顶Color Match 颜色匹配(Curve Editor)Control 控制Node 节点Insert 插入Snap 快照Guidelines 参考线Ruler Grid 标尺格Smooth 平滑(Transducer Model Derivation)Measurements 度量Profile 配置文件(Scale Parameters)Axis 轴Range 范围Divisions 分格Current 当前Volume 音量Acceleration 加速度Velocity 速率Excursion 偏移。
基于LMS算法和Matlab的自适应滤波器的设计
第29卷第4期2008年8月华 北 水 利 水 电 学 院 学 报Journal of North China I nstitute of W ater Conservancy and Hydr oelectric PowerVol 129No 14Aug .2008收稿日期:2008-04-30作者简介:陈黎霞(1979—),女,河南周口人,助教,硕士,主要从事信号检测与处理方面的研究.文章编号:1002-5634(2008)04-0051-03基于LM S 算法和M a tl ab 的自适应滤波器的设计陈黎霞,李亚萍,姚淑霞(华北水利水电学院,河南郑州450011)摘 要:根据LMS 算法性能特点,在Matlab 环境下编写了基于L M S 算法的有限长自适应滤波器的程序(3.m ),用所设计的滤波器对受白噪声干扰的语音信号及正弦波信号进行滤波.理论分析和仿真结果表明,所设计的自适应滤波器具有快速的跟踪能力和收敛性能,且稳态误差较小.关键词:LMS 算法;自适应滤波器;Matlab 中图分类号:T N273.2 文献标识码:A 自适应滤波器在信号处理领域占有极其重要的地位,广泛应用于通信、雷达、导航系统和工业控制等方面.在一些无法预知信号和噪声特性的场合,无法使用具有固定滤波器系数的滤波器对信号实现最优滤波,其惟一的解决办法是引入自适应滤波器.使用Matlab 的信号处理功能及工具箱能够快速有效地实现自适应滤波器的分析、设计及仿真,在设计中可以随时更改参数,以达到滤波器设计的最优化,节约开发时间.1 基于LMS 算法的自适应滤波原理自适应滤波器与普通滤波器的区别是它能够随着外界信号特性动态地改变参数,保持最佳滤波状态.如何根据外界信号的变化来调整参数是由自适应算法决定的,因此自适应算法的好坏直接影响滤波的效果[1].L MS 算法是利用梯度估计值来代替梯度向量的一种快速搜索算法,具有计算量小、易实现的优点[2];其基本思想是通过调整滤波器的权值参数,使滤波器的输出信号与期望信号之间的均方误差最小.自适应滤波一般包括2个基本过程:滤波过程和滤波器参数调整过程.这2个过程组成1个反馈环,如图1所示,x (n )为输入信号,y (n )为输出信号,d (n )为参考信号,e (n )为y (n )与d (n )的误差信号.自适应滤波器的滤波系数受误差信号e (n )控制.虽然F I R 和II R 结构都可以用于自适应滤波器,由于II R 的稳定性问题,所以设计时采用自适应横向F I R 滤波器,其结构如图2所示[1].图1 自适应滤波器原理图图2 基于LM S 算法的自适应横向F I R 滤波器原理图 在滤波过程中其输入为X (n )=[x (n ),x (n -1),…,x (n -N +1)]TN 阶F I R 滤波器的权系数W (n )=[w 1(n ),w 2(n ),…,w N (n )]T滤波器的输出y (n )=∑Ni =1w i x (n -i +1)=X T(n )W (n )估计误差为e (n )=d (n )-y (n )=d (n )-X T(n )W (n )根据最小均方误差准则,最佳滤波器参量应使得性能函数———均方误差E [e 2(n )]为最小,因此自适应滤波的优化准则为m in∑nk =1(e (k ))2=m in∑nk =1(d (k )-y (k ))2估计误差与输入向量都被加到自适应控制部分,可以采用最优化方法中的最速下降法求自适应滤波器的最佳权向量[2]W (n +1)=W (n )+2μe (n )X (n )式中:μ为步长因子,是控制收敛速度和稳态误差的参量,且0<μ<1/λmax ;λmax 为输入信号自相关矩阵的最大特征值[3].当选择合适的步长因子μ,采用瞬时输出误差功率的梯度作为均方误差梯度的估计时,可使均方误差E [e 2(n )]趋于最小值,因此加快滤波器的收敛速度可以从滤波器长度控制和步长因子控制着手.2 Matlab 仿真设计2.1 语音通信中噪声消除仿真在通信中,语音信号不可避免地受到周围环境、通信设备内部电噪声等因素的影响,使接受者收到的语音信号被污染,影响通话质量,因此在语音通信中滤波是必不可少的过程[4].考虑到实际通信中噪声频谱很宽,因此在仿真中对纯净语音的干扰设置为均值为零的高斯白噪声,为加速Matlab 的运行速度,在程序设计时使用向量和矩阵化的算法[5].信噪比不同时所需滤波器阶数和步长也不同,图3和图4分别是信噪比为0.1dB 时的语音信号时域滤波效果及滤波器学习曲线图,自适应滤波器长度为50,计算出最大步长因子μmax =0.6157,选取μ=0.0040.从图3中可观察到滤波后的信号逼近原始语音信号,且通过回放的语音可证实这一点.部分仿真程序如下:…;[x,fs]=wavread (’D:\hy .wav ’,[1654043080]);number =length (x );t =0:1/fs:(size (x )-1)/fs;noise =[wgn (1,length (x ),-15)]’;Sn =x +noise;tic L =50;b =fir1(L -1,0.5);mu max =maxstep (ha,x );…;wn =adap tfilt .l m s (L,mu );m =10;m se =m sesi m (wn,x,d,m );t =t oc;…;s ound (ys );…2.2 正弦波信号除噪图5是用该自适应滤波器对含随机噪声单频正弦波信号滤波处理的效果.滤波器长度30,步长因25 华 北 水 利 水 电 学 院 学 报 2008年8月子μ=0.0020.从图5中看出该滤波器能完成除噪.图6为自适应滤波器学习曲线,其收敛速度较快.3 仿真结果分析1.在信噪比较大时,L MS 自适应滤波器滤波结果很好,要求的滤波器长度也较短,收敛速度较快.2.在信噪比较小时,自适应滤波器的输出结果不十分理想,但可以通过适当调整步长参数及适当增加滤波器长度来进行改进,且效果较明显.3.自适应滤波器的收敛速度在很大程度上取决于步长因子μ.当步长参数较大时,滤波器收敛到稳态需要迭代次数较少,但滤波效果比μ较小时差,而且均方误差的稳态值随着μ的变大而增大;但是当步长参数较小时,收敛速度则会降低,因此只有选择合适的步长参数μ,才能使该滤波器的性能稳定.4 结 语仿真结果显示,采用基于L MS 算法设计出的自适应滤波器有良好的收敛性、较小的稳态误差,噪声功率较大的情况下也能完成数字滤波任务,在噪声消除方面具有很好的效果和性能.由于Matlab 具有强大的接口功能,仿真后的结果可以很方便地移植到数字信号处理器、可编程逻辑器件等中,为自适应滤波器的硬件实现打下了良好的理论基础.参 考 文 献[1]张贤达.现代信号处理(第二版)[M ].北京:清华大学出版社,2002.[2]邹艳碧,高鹰.自适应滤波算法综述[J ].广州大学学报,2007,1(2):44-48.[3]黄振远,朱剑平.自适应滤波LMS 类算法探究[J ].现代电子技术,2006,24:52-54.[4]编著责任者不详.离散语音信号处理[M ].赵胜辉,译.北京:电子工业出版社,2004.[5]钟麟.Matlab 仿真技术与应用教程[M ].北京:国防工业出版社,2004.[6]丁元力.M atlab 语言在数字语音处理上的应用[J ].电声技术,2001,25(9):7-9.D esi gn of Adapti ve F ilter Ba sed on LM S A lgor ithm i n M a tl abCHE N L i 2xia,L I Ya 2p ing,Y AO Shu 2xia(North China I nstitute of W ater Conservancy and Hydr oelectric Power,Zhengzhou 450011,China )Abstract:According t o the character of L M S algorith m,the p r ogram s (3.m )of finite adap tive filter based on LMS in Matlab is com 2p iled,and the designed filter is used t o accomp lish the filter of audi o signal and sine wave which are disturbed by white noise .The con 2clusi on fr om the theoretic analysis and si m ulati on result show that,the designed filter has fast track and convergency capability al ong with s mall steady 2state err or .Key words:L M S algorith m;Adap tive Filter;Matlab35第29卷第4期陈黎霞等: 基于L M S 算法和M atlab 的自适应滤波器的设计 。
LMS中英文对照表
LMS 中英文对照表位置原文简体中文繁体中文菜单File 文件New 新建Open 打开Reopen 重新打开Save 保存SaveAs 另存为Revert 返回Load QuickSet File 载入快速设置文件Save QuickSet File夹保存快速设置文件Load GraphSetup File 载入图表设置文件Save GraphSetup File 保存图表设置文件Print 打印Editor 编辑器Preferences 参数选择Exit 退出Graph 图表Parameters 参数Curve Library 曲线库Notes & Comments 附注&注释Analyzer 分析仪Sweep Start/Stop 扫频开始/停止Osc On/Off Osc 开/关RLC meter RLC 表Microphone Setup 麦克风设置PAC Interface PAC接口Micro Run 宏运行Calibration 剪贴板Processing 处理Unary Math Operations 一元数学操作Binary Math Operations 二元数学操作Minimum Phase Transform 最小相位转换Delay Phase Transform 延迟相位转换Group Delay Transform 组延迟转换Inv Fast Fourier Transform 快速反傅立叶转换Fast Fourier Transform 快速傅立叶转换Speaker Parameters 喇叭参数Tail Correction 尾部修正Data Transfer 数据合并Data Splice 数据接合Data Realign 数据重组Curve Averaging 数据提升Curve Compare 数据对照Curve Integration 数据综合Utilities 实用程序Import Curve Data File 导入曲线数据文件Export Curve Data File 导出曲线数据文件Export Graphics to File 导出图表到文件Export Graphics to Clipboard 导出图表到剪贴板Curve Capture 曲线捕捉Curve Editor 曲线编辑器Macro Editor 宏编辑器MDF Editor MDF 编辑器Polar Convertor 极性转换器View Clipboard 查看剪贴板Transducer Model Derivation 传感器模式导出Scale 刻度Parameters 参数Auto 自动Up 向上Down 向下View 查看Zoom In 放大Zoom Out 缩小Zoom 缩放Redraw 刷新ToolBars 工具栏Show All 全部显示Hide All 全部隐藏ToolBox 工具框Help 帮助Contents 内容Index 索引Glossary 术语表About Modules 关于模块About Program 关于软件对话框(Graph Parameters)Frame 边框Background 背景Note Underline 注释下划线Large Frame Line 大边框线Small Frame Line 小边框线Grid 格子Border Line 边界线Major Div 主格Minor Div 次格Font 字体Title Block 工程明细表Map Legend 映射图例Note List 注释列表Graph Title 图表标题Scale Vertical 垂直刻度Scale Horizontal 水平刻度Typeface 字体Style 字形Size 大小Color 颜色(Curve Library)Data Curve 数据曲线Same Line Type 统一线型Left(Magnitude) 左坐标(量级) Right Lighter 右坐标浅色Right(Phase) 右坐标(相位)Info 信息Horz Data Range 频宽范围Left Vert 左坐标Right Vert 右坐标Points 点Style 类型Width 宽度Color 颜色(Note&Comments)Left Page 左页面Right Page 右页面Title Block Data 工程图明细表Person 个人Company 公司Project 工程Automatic Curve Info Notes 自动曲线信息标注(Analyzer Parameters)Oscillator 振荡器Output Level 输出水平Frequency 频率Mode 扫频模式Hi Speed Data 高速数据Precision Data 精密数据Gating 门控Off 关On 开Meter 仪表Data 数据Value 值Source 来源Freq 频率Sweep 扫频Lo Freq 低频Hi Freq 高频Direction 方向Control 控制Pulse 脉冲Gate Time Calculator 门控计时器Meter Filter 滤波器Filter Function 滤波函数Track Ratio 音轨率Gate Timing 门控时间(RLC Meter)Measurement 测量Resistance 电阻Inductance 电感Capacitance 电容Impedance 阻抗Limit Testing 测试限制Enable 启用MinValue 最小值Max Value 最大值(Microphone Setup)Mic Input 麦克风输入Line Input 信号输入Model 型号Acoustic Ref 声学参数Electric Ref 电学参数Serial 序列号Author 制造商Date 日期Load 载入(PAC Interface)Serial Port 串行端口Linking 正在链接Start Link 开始链接Automatic Link 自动链接Baud Rate 波特率System Power 系统供电Power Source 供电电源Battery Status 电池状态Charge Amps 功放充电System Status 系统状态Link Status 链接状态Port 端口Voltage 电压Battery V oltage 电池电压External V oltage 外部电压(Analyzer Calibration)Internal 内部External 外部Parameter Under Test 测试参数Results 结果(Unary Math Operations)Library Curve 曲线库Operation 操作Magnitude Offset 幅度补偿Phase Offset 相位偏移Delay Offset 延迟偏移Exponentiation 求幂Smooth Curve 平滑曲线Frequency Translation 频率转换Multiply by 乘Divide 除Real 正弦Imag(sin) 余弦Execute 执行(Binary Math Operations)Mul 乘Div 除Add 加Sub 减Operand 操作数(Minimum Phase Transform)Asymptotic Slope at Hi Freq Limit 频率上限斜率Asymptotic Slope at Lo Freq Limit 频率下限斜率Automatic Tail Correction/Mirroring for Impedance Curves 自动尾部修正/阻抗曲线镜像(Delay Phase Transform)Source Curve with Delay Data 来源延迟数据曲线Result Curve for Phase Data 结果相位数据曲线(Group Delay Transform)Source Curve with Phase Data 来源相位曲线Result Curve for Group Delay 结果组延迟曲线(Inverse Fast Fourier Transform)Linear Frequency Points 线性频率点Frequency Domain Data 频率范围数据Result Impulse Curve 结果推动曲线Time Domain Data 时间范围数据Result Step Curve 结果步骤曲线(Speaker Parameters)Method 方法Single Curve 单曲线Double Curve 双曲线Reference Curve 参考曲线Standard 标准Estimate 估算Optimize 优化Simulate 模拟Model Simulation 模拟模型Copy Binary to Clipboard 复制二元数据到剪贴板Copy Text to Clipboard 复制文本到剪贴板(Tail Correction)Slope Hi 高频Slope Lo 低频(Data Splice)Source Curve for Higher Data 来源高数据曲线Source Curve for Lower Data 来源低数据曲线Horz Splice Transition 水平接合转换(Data Realign)Linear 线性Log 对数Horz Lo Limit 水平下限Horz Hi Limit 水平上限Interpolation 插补Quadratic 平方Cubic 立方(Curve Averaging)Scalar 标量Vector 矢量(Curve Compare)Test Parameters 测试参数Absolute 绝对Relative 相对Tolerance 公差Relative Flatness 相对平面(Curve Integration)Average 平均Integrate 叠加(Import Curve Data)Left Vert Data 左坐标数据Right Ver Data 右坐标数据Polar Freq 极性频率Units 单位Prefix 前缀Curve Entry 曲线条目Special Processing 特殊处理Skip First Data Column 跳过首个数据列(Export Graphics)Artwork 作品Graph Name 图表名称Format 格式Raster 光栅Image Pixel Width 图像宽度(像素) Image Pixel Height 图像高度(像素) Image Bits per Pixels 图像位每像素Image Bytes 图像字节数(Clipboard Graphics Transfer)Enhanced Metafile 增强元文件Bitmap 位图(Curve Capture)Polar Freq 极性频率Graph Image 图表图像Reference Point Data 参考点数据Upper Right 右上Scan Direction 扫描方向Top to Bottom 顶到底Bottom to Top 底到顶Color Match 颜色匹配(Curve Editor)Control 控制Node 节点Insert 插入Snap 快照Guidelines 参考线Ruler Grid 标尺格Smooth 平滑(Transducer Model Derivation)Measurements 度量Profile 配置文件(Scale Parameters)Axis 轴Range 范围Divisions 分格Current 当前Volume 音量Acceleration 加速度Velocity 速率Excursion 偏移。
基于LMS算法与RLS算法的自适应滤波
基于LMS算法与RLS算法的自适应滤波徐艳;李静【期刊名称】《电子设计工程》【年(卷),期】2012(020)012【摘要】The theory and technology of adaptive signal processing have become popular in filtering and canceling noise field. This article mainly talks about the theory of adaptive filtering and steps of the arithmetic based on LMS&RLS.Emulations successfully showed the theory by MATLAB in this paper.%自适应信号处理的理论和技术已经成为人们常用滤波和去噪技术。
文中讲述了自适应滤波的原理以及LMS算法和RLS算法两种基本自适应算法的原理及步骤。
并用MATLAB分别对两种算法进行了自适应滤波仿真和实现。
【总页数】4页(P49-51,54)【作者】徐艳;李静【作者单位】长安大学信息工程学院,陕西西安710064;长安大学信息工程学院,陕西西安710064【正文语种】中文【中图分类】TP312【相关文献】1.基于LMS算法与RLS算法自适应滤波及仿真分析 [J], 马国栋;阎树田;贺成柱;杨晨2.基于RLS算法的自适应抗干扰工频通信滤波器的设计与实现 [J], 徐婷;通旭明3.基于qF-LMS算法的自适应滤波器与FPGA实现 [J], 丁泽锋; 白路阳; 杨炜毅; 王艳芬4.基于LMS算法的自适应滤波器性能分析 [J], 刘建涛;席闯;姜海洋5.基于RLS算法的有源滤波器自适应基波检测方法(英文) [J], 姜孝华;金济;Ale Emedi因版权原因,仅展示原文概要,查看原文内容请购买。
外文翻译---基于LMS自适应滤波器在直达波消除中的运用
外文翻译学生姓名学号院系电子与信息工程专业电子信息工程指导教师二O一一年六月二日基于LMS 自适应滤波器在直达波消除中的运用徐元军,陶源,王越,单涛电子工程系,信息科学与技术学院,北京理工大学,北京100081,中国摘要:本文介绍了使用最小均方(LMS)算法消除无源雷达收到的直达波。
并由此推导出直达波的模型。
通过使用基于LMS 算法的FIR 自适应滤波器,从而开发出来调频无源雷达的软件解决方案,从而代替了利用硬件对无源雷达的调试。
由此我们获得的一些无源雷达的仿真结果。
这些仿真结果预示着利用LMS 算法消除直达波是十分有效的。
关键字:LMS 算法;自适应滤波器;直达波消除;在以往的雷达系统的研究中,大多数的雷达专家都曾经专注于无源雷达系统,但是只是把它当做只用作为商业电台的广播电台发射器,比如电视和GSM 发射机等。
而这种无源雷达系统的其他的一些潜在运用仅仅只是在一些实验[1]中被介绍.无源雷达系统通常包括一个参考接收器和一个回波接收器。
在实际中,无源雷达的回波接收器通常不仅收到目标的回波,而且也接收到由于多径传播效应而产生的回波。
由于在实际中的雷达的横截面(RCS)的目标通常是非常小的,与多径传播效应而产生的回波相比,目标的回波是非常微弱的,这使得检测信号变得十分困难。
这就是为什么在这种情况下,实现目标的检测成为一项极其艰巨的任务。
在实际中,无源雷达设备使用了各种各样的不同方案来解决这个问题[2][3]。
但是这些方法都需要添加特殊的硬件才能够实现直达波的消除。
为了解决这个问题,现在我们可以采用软件的方法来实现直达波的消除。
在过去几十年的滤波器理论研究中,自适应信号处理经过不断的发展已经成为了现在研究的热门领域之一。
越来越多的自适应理论被广泛地运用于实际生活和生产中。
实际中的一些重要的运用主要包括自适应线性预测,回波消除,自适应通道均衡等。
自适应理论的这些运用使我们意识到也可以采用自适应滤波器来实现直达波消除。
神经网络基于LMS算法的自适应滤波
实验报告实验名称:基于LMS算法的自适应滤波实验报告实验名称:基于LMS算法的自适应滤波实验内容:最小均方算法即LMS是一种自适应滤波算法,这里的Matlab程序根据LMS对一个线性噪声系统进行滤波。
实验原理:最小均方算法是一种以期望响应和滤波器输出信号之间误差的均方值最小为原则,依据输入信号在迭代过程中估计梯度矢量,并更新权系数以达到最佳的自适应迭代算法。
实验程序:clear;clc;grid off;%周期信号的产生t=0:99;sn=10*sin(0.5*t);figure(1)subplot(2,1,1)plot(t,sn);title('原始的周期信号')grid on;%噪声信号的产生randn('state',sum(100*clock));noise=randn(1,100);subplot(2,1,2)plot(t,noise);title('噪声信号')grid on;%信号滤波xn=sn+noise;xn=xn.';dn=sn.';M=20;figure(2)subplot(2,1,1)plot(t,xn)title('加噪声后的信号波形')grid on;%初始化r_max=max(eig(xn*xn.'));mu=rand()/r_max;itr=length(xn);en=zeros(itr,1);w=zeros(M,itr);%LMSfor k=M:itrx=xn(k:-1:k-M+1);y=w(:,k-1).'*x;en(k)=dn(k)-y;%加权因子w(:,k)=w(:,k-1)+2*mu*en(k)*x;endyn=inf*ones(size(xn));for k=M:itrx=xn(k:-1:k-M+1);yn(k)=w(:,end).'*x;end%画图subplot(2,1,2)plot(t,yn)title('滤波器输出信号')grid on;figure(3)hold on;plot(t,dn,'g',t,yn,'b',t,dn-yn,'r'); grid on;legend('期望输出','滤波器输出','误差')实验结果仿真:实验总结:LMS算法是一种梯度最速下降算法,其显著特点是简单、计算量小、易于实现。
基于LMS算法的自适应滤波器研究
基于LMS算法的自适应滤波器研究摘要自适应滤波器是一种现代滤波器,自适应滤波器也是相对于固定滤波器来说的。
而固定滤波器是一种经典滤波器,其滤波频率是固定不变的,自适应滤波器的滤波频率则是跟随自动适应输入信号而改变的,其适用范围更加广泛。
在没有任何关于信号和噪声的先验知识的条件下,其自适应体现在:采用前一时刻已获得的滤波器参数来自动调节现在时刻的滤波器参数。
因此,能够适应信号和噪声未知或者随机变化的未知特性,从而实现了最优滤波。
关键词LMS算法;自适应滤波器1 LMS自适应算法自适应滤波器在实质上就是一种能调节其自身传输特性以达到最优化的维纳滤波器。
自适应滤波器除了包括一种按结构设计的滤波器,还有一种自适应的算法[1]。
自适应滤波器的算法主要是以各种判断来设计完成的,以各种判据条件作为推算的基础。
通常有两种判据条件:最小均方误差判据和最小二乘法判据。
LMS算法是以最小均方误差为判据的最典型的算法,也是应用最广泛的一种算法。
令y(n)为输入x(n)时神经元k在n时刻的实际输出,d(n)表示期望的输出(可由训练样本给出),则误差信号可写为:2 LMS自适应滤波器仿真為验证LMS算法构成的自适应滤波器的实际效果,特做以下仿真实验:如图1所示,原始信号为正弦波形,在0s时对系统原始信号加入一个干扰信号,在经过加噪后波形严重失调。
如图2所示,在时间1s时使LMS自适应滤波器开始工作。
由上图可知在LMS自适应滤波器开始工作后,原本已经严重失调的原始信号能够立即完成扰动消除,并且逐渐向正常波形恢复。
3 结束语LMS算法虽然是自适应滤波器应用最为广泛的算法。
但其形成的自适应滤波器具有良好的消除扰动效果,并通过仿真实验来证明。
参考文献[1] 王万召,王杰.过热汽温自适应逆控制方案研究[J].电力自动化设备,2013,33(9):54-57.。
自适应滤波:最小均方误差滤波器(LMS、NLMS)
⾃适应滤波:最⼩均⽅误差滤波器(LMS、NLMS)作者:桂。
时间:2017-04-02 08:08:31链接:声明:欢迎被转载,不过记得注明出处哦~【读书笔记08】前⾔西蒙.赫⾦的《⾃适应滤波器原理》第四版第五、六章:最⼩均⽅⾃适应滤波器(LMS,Least Mean Square)以及归⼀化最⼩均⽅⾃适应滤波器(NLMS,Normalized Least Mean Square)。
全⽂包括: 1)LMS与维纳滤波器(Wiener Filter)的区别; 2)LMS原理及推导; 3)NLMS推导; 4)应⽤实例;内容为⾃⼰的读书记录,其中错误之处,还请各位帮忙指出!⼀、LMS与维纳滤波器(Wiener Filter)的区别这⾥介绍的LMS/NLMS,通常逐点处理,对应思路是:随机梯度下降;对于Wiener Filter,给定准则函数J,随机/批量梯度都可以得出最优解;LMS虽然基于梯度下降,但准则仅仅是统计意义且通常引⼊误差,可以定义为J_0,简⽽⾔之J通常不等于J_0,得出的最优解w_o⾃然也通常不等于维纳最优解;分析LMS通常会分析稳定性,稳定性是基于Wiener解,。
但LMS是Wiener解的近似,所以:迭代步长的稳定性,严格适⽤于Wiener 解,对于LMS只是⼀种近似参考,并没有充分的理论依据。
下⽂的分析仍然随机梯度下降的思路进⾏。
⼆、LMS原理及推导LMS是时间换空间的应⽤,如果迭代步长过⼤,仍然有不收敛的问题;如果迭代步长过⼩,对于不平稳信号,还没有实现寻优就⼜引⼊了新的误差,屋漏偏逢连夜⾬!所以LMS系统是脆弱的,信号尽量平稳、哪怕短时平稳也凑合呢。
给出框图:关于随机梯度下降,可以。
这⾥直接给出定义式:利⽤梯度下降:- \nabla J = {\bf{x}}{\left( {{{\bf{w}}^T}{\bf{x}} - {d}} \right)^T}给出LMS算法步骤:1)给定\bf{w}(0),且1<\mu<1/\lambda_{max};2)计算输出值:y\left( k \right) = {\bf{w}}{\left( k \right)^T}{\bf{x}}\left( k \right);3)计算估计误差:e\left( k \right) = d\left( k \right) - y\left( k \right);4)权重更新:{\bf{w}}\left( {k + 1} \right) = {\bf{w}}\left( k \right) + \mu e\left( k \right){\bf{x}}\left( k \right)三、NLMS推导看到Normalized,与之联系的通常是约束条件,看到约束不免想起拉格朗⽇乘⼦。
外文翻译---基于LMS自适应滤波器在直达波消除中的运用
外文翻译---基于LMS自适应滤波器在直达波消除中的运用本文介绍了使用最小均方(LMS)算法消除无源雷达收到的直达波,并推导出直达波的模型。
通过使用基于LMS算法的FIR自适应滤波器,开发出了调频无源雷达的软件解决方案,代替了利用硬件对无源雷达的调试。
仿真结果表明,利用LMS算法消除直达波是十分有效的。
在以往的雷达系统研究中,无源雷达系统被认为只能用于商业电台的广播电台发射器,如电视和GSM发射机等。
而无源雷达系统的其他潜在应用仅在一些实验中被介绍。
实际中,无源雷达的回波接收器通常不仅收到目标的回波,还接收到由于多径传播效应而产生的回波。
由于目标的回波非常微弱,检测信号变得十分困难。
因此,实现目标的检测成为一项艰巨的任务。
为解决这个问题,可以采用软件的方法实现直达波的消除。
自适应信号处理是滤波器理论研究中的热门领域之一,越来越多的自适应理论被广泛地运用于实际生活和生产中,如自适应线性预测、回波消除、自适应通道均衡等。
通过分析直达波的特性,可以采用自适应滤波器来实现直达波消除。
基于LMS自适应滤波器可以被用来解决这个问题。
本文的主要内容是介绍使用LMS算法消除无源雷达收到的直达波,并推导出直达波的模型。
通过使用基于LMS算法的FIR自适应滤波器,开发出了调频无源雷达的软件解决方案,代替了利用硬件对无源雷达的调试。
仿真结果表明,利用LMS算法消除直达波是十分有效的。
为了分析直达波消除问题,需要建立一个准确的直达波模型。
通过比较,发现直达波与无线电信道中的多径传播非常相似,都由不同延迟的信号振幅构成。
因此,可以用无线电信道系统中的多径传播模型来表示直达波的脉冲响应。
直达波的脉冲响应可表示为一个连续时间FIR滤波器的形式,其中振幅、时间延迟和相移是信号多径传播的总路数。
在无源雷达系统中,接收器输出与数字信号处理器相连。
引入复杂参数ai替换原公式中的an,对公式进行Z变换,得到直达波在Z域的模型表达式。
将其看作FIR滤波器的传递函数,通过已知的脉冲响应,可以对直达波进行估计。
基于LMS算法的自适应组合滤波器中英文翻译
Combined Adaptive Filter with LMS-Based Algorithms ´Abstract: A combined adaptive filter is proposed. It consists of parallel LMS-based adaptive FIR filters and an algorithm for choosing the better among them. As a criterion for comparison of the considered algorithms in the proposed filter, we take the ratio between bias and variance of the weighting coefficients. Simulations results confirm the advantages of the proposed adaptive filter.Keywords: Adaptive filter, LMS algorithm, Combined algorithm,Bias and vari ance trade-off1.IntroductionAdaptive filters have been applied in signal processing and control, as well as in many practical problems, [1, 2]. Performance of an adaptive filter depends mainly on the algorithm used for updating the filter weighting coefficie nts. The most commonly used adaptive systems are those based on the Least Mean Square (LMS) adaptive algorithm and its modifications (LMS-based algorithms).The LMS is simple for implementation and robust in a number of applications [1–3]. However, since it does not always converge in an acceptable manner, there have been many attempts to improve its performance by the appropriate modifications: sign algorithm (SA) [8], geometric mean LMS (GLMS) [5], variable step-size LMS(VS LMS) [6, 7].Each of the LMS-base d algorithms has at least one parameter that should be defined prior to the adaptation procedure (step for LMS and SA; step and smoothing coefficients for GLMS; various parameters affecting the step for VS LMS). These parameters crucially influence the filter output during two adaptation phases:transient and steady state. Choice of these parameters is mostly based on some kind of trade-off between the quality of algorithm performance in the mentioned adaptation phases. We propose a possible approach for the LMS-based adaptive filter performance improvement. Namely, we make a combination of several LMS-based FIR filters with different parameters, and provide the criterion for choosing the most suitable algorithm for different adaptation phases. This method may be applied to all the LMS-based algorithms, although we here consider only several of them.The paper is organized as follows. An overview of the considered LMS-based algorithms is given in Section 2.Section 3 proposes the criterion for evaluation and combination of adaptive algorithms. Simulation results are presented in Section 4.2. LMS based algorithms Let us define the input signal vector T k N k x k x k x X )]1()1()([+--= and vector of weighting coefficients as T N k k W k W k W W )]()()([110-= .The weighting coefficients vector should be calculated according to:}{21k k k k X e E W W μ+=+ (1) where µ is the algorithm step, E{·} is the estimate of the expected value and k T k k k X W d e -=is the error at the in-stant k,and dk is a reference signal. Depending on the estimation of expected value in (1), one defines various forms of adaptive algorithms: the LMS {}()k k k k X e X e E =,the GLMS {}()()∑=--≤<-=k i i k i k i k k a X e a a X e E 010,1, and the SA {}()()k k k k e sign X X e E =,[1,2,5,8] .The VS LMS has the same form as the LMS, but in the adaptation the step µ(k) is changed [6, 7].The considered adaptive filtering p roblem consists in trying to adjust a set of weighting coefficients so that the system output,k T k k X W y =, tracks a reference signal, assumed as k k Tk k n X W d +=*,where k n is a zero mean Gaussian noise with thevariance 2n σ,and *k W is the optimal weight vector (Wiener vector). Two cases will be considered:W W k =* is a constant (stationary case) and *k W is time-varying (nonstationary case). In nonstationary case the unknown system parameters( i.e. the optimal vector *k W )are time variant. It is often assumed that variation of *k W may be modeled as K k k Z W W +=+**1 is the zero-mean random perturbation, independent onk X and k n with the autocorrelation matrix []I Z Z E G Z T k k 2σ==.Note that analysis for the stationary case directly follows for 02=Zσ.The weighting coefficient vector converges to the Wiene r one, if the condition from [1, 2] is satisfied.Define the weighting coefficientsmisalignment, [1–3],*k k k W W V -=. It is due to both the effects of gradient noise (weighting coefficients variations around the average value) and the weighting vector lag (difference between the average and the optimal value), [3]. It can be expressed as:()()()()*k k k k k W W E W E W V -+-=, (2)According to (2), the ith element of k V is:(3)where ()()k W bias i is the weighting coefficient bias and ()k i ρ is a zero-mean random variable with the variance 2σ.The variance depends on the type ofLMS-based algorithm, as well as on the external noise variance 2n σ.Thus, if the noisevariance is constant or slowly-varying,2σ is time invariant for a particular LMS-based algorithm. In that sense, in the analysis that follows we will assume that 2σ depends only on the algorithm type, i.e. on its parameters.An important performance measure for an adaptive filter is its mean square deviation (MSD) of weighting coefficients. For the adaptive filters, it is given by, [3]:[]k T k k V V E MSD ∞→=lim .3. Combined adaptive filterThe basic idea of the combined adaptive filter lies in parallel implementation of two or more adaptive LMS-based algorithms, with the choice of the best among them in each iteration [9]. Choice of the most appropriate algorithm, in each iteration, reduces to the choice of the best value for the weighting coefficients. The best weighting coefficient is the one that is, at a given instant, the closest to the corresponding value of the Wiener vector.Let ()q k W i , be the i −th weighting coefficient for LMS -based algorithm with the chosen parameter q at an instant k. Note that one may now treat all the algorithms in a unified way (LMS: q ≡ µ,GLMS: q ≡ a,SA:q ≡ µ). LMS -based algorithm behavior is crucially dependent on q. In each iteration there is an optimal value qopt , producing the best performance of the adaptive al-gorithm. Analyze no w a combined adaptive filter, with several LMS -based algorithms of the same type, but with different parameter q.The weighting coefficients are random variables distributed around the ()k W i *,with()()q k W bias i ,and the variance 2q σ, related by [4, 9]: ()()()()q i i i q k W bias k W q k W κσ≤--,,*, (4)where (4) holds with the probability P(κ), dependent on κ. For example, for κ = 2 and ()()()()()()()()()()()()k k W bias k W E k W k W k W E k V i i i i i i i ρ+=-+-=*a Gaussian distribution,P(κ) = 0.95 (two sigma rule).Define the confidence intervals for ()]9,4[,,q k W i :()()()[]q i q i i q k W k q k W k D κσσ2,,2,+-= (5) Then, from (4) and (5) we conclude that, as long as ()()q i q k W bias κσ<,,()()k D k W i i ∈*, independently on q. This means that, for small b ias, the confidence intervals, for different s q ' of the same LMS-based algorithm, of the same LMS-based algorithm, intersect. When, on the other hand, the bias becomes large, then the central positions of the intervals for different s q ' are far apart, and they do not intersect.Since we do not have apriori information about the ()()q k W bias i ,,we will use a specific statistical approach to get the criterion for the choice of adaptive algorithm, i.e. for the values of q. The criterion follows from the trade-off condition that bias and variance are of the same order of magnitude, i.e.()()[]4,,q i q k W bias κσ≅.The proposed combined algorithm (CA) can now be summarized in the following steps:Step 1. Calculate ()q k W i ,for the algorithms with different s q 'from the predefined set {} ,,2q q Q i =.Step 2. Estimate the variance 2q σ for each considered algorithm.Step 3. Check if ()k D i intersect for the considered algorithms. Start from an algorithm with largest value of variance, and go toward the ones with smaller values of variances. According to (4), (5) and the trade-off criterion, this check reduces to the check if()()()ql qm l i m i q k W q k W σσκ+<-2,, (6)is satisfied, where Q q q l m ∈,,and the following relation holds:Q q q h ql qh qm h ∉⇒>>∀,:222σσσ.If no ()k D i intersect (large bias) choose the algorithm with largest value of variance. If the ()k D i intersect, the bias is already small. So, check a new pair of weighting coefficients or, if that is the last pair, just choose the algorithm with the smallest variance. First two intervals that do not intersect mean that the proposed trade-off criterion is achieved, and choose the algorithm with large variance.Step 4. Go to the next instant of time.The smallest number of elements of the set Q is L =2. In that case, one of the s q 'should provide good tracking of rapid variations (the largest variance), while the other should provide small variance in the steady state. Observe that by adding few more s q ' between these two extremes, one may slightly improve the transient behavior of the algorithm.Note that the only unknown values in (6) are the variances. In our simulations weestimate 2q σ as in [4]:()()()2675.0/1--=k W k W median i i q σ, (7)for k = 1, 2,... , L and 22qZ σσ<<. The alternative way is to estimate 2n σ as:∑=≈T i i n e T 1221σ,for x(i) = 0. (8) Expressions relating 2n σ and 2q σ in steady state, for different types of LMS-basedalgorithms, are known from literature. For the standard LMS algorithm in steady state, 2n σ and 2q σ are related 22nq q σσ=,[3]. Note that any other estimation of 2q σis valid for the proposed filter.Complexity of the CA depends on the constituent algorithms (Step 1), and on the decision algorithm (Step 3).Calculation of weighting coefficients for parallel algorithms does not increase the calculation time, since it is performed by a parallel hardware realization, thus increasing the hardware requirements. The variance estimations (Step 2), negligibly contribute to the increase of algorithm complexity, because they are performed at the very beginning of adaptation and they are using separate hardware realizations. Simple analysis shows that the CA increases the number of operations for, at most, N(L −1) additions and N(L −1) IF decisions, and needs some additional hardware with respect to the constituent algorithms.4.Illustration of combined adaptive filterConsider a system identification by the combination of two LMS algorithms with different steps. Here, the parameter q is μ,i.e. {}{}10/,,21μμ==q q Q .The unknown system has four time-invariant coefficients,and the FIR filters are with N = 4. We give the average mean square deviation (AMSD ) for both individual algorithms, as well as for their combination,Fig. 1(a). Results are obtained by averaging over 100 independent runs (the Monte Carlo method), with μ = 0.1. Thereference dk is corrupted by a zero-mean uncorrelated Gaussian noise with 2n σ= 0.01and SNR = 15 dB, and κ is 1.75. In the first 30 iterations the variance was estimated according to (7), and the CA picked the weighting coefficients calculated by the LMS with μ.As presented in Fig. 1(a), the CA first uses the LMS with μ and then, in the steady state, the LMS with μ/10. Note the region, between the 200th and 400th iteration,where the algorithm can take the LMS with either stepsize,in different realizations. Here, performance of the CA would be improved by increasing the number of parallel LMS algorithms with steps between these two extrems.Observe also that, in steady state, the CA does not ideally pick up the LMS with smaller step. The reason is in the statistical nature of the approach.Combined adaptive filter achieves even better performance if the individual algorithms, instead of starting an iteration with the coefficient values taken from their previous iteration, take the ones chosen by the CA. Namely, if the CA chooses, in the k -th iteration, the weighting coefficient vector P W ,then each individual algorithm calculates its weighting coefficients in the (k +1)-th iteration according to:{}k k p k X e E W W μ21+=+(9)Fig. 1. Average MSD for considered algorithms.Fig. 2. Average MSD for considered algorithms.Fig. 1(b) shows this improvement, applied on the previous example. In order to clearly compare the obtained results,for each simulation we calculated the AMSD . For the first LMS (μ) it was AMSD = 0.02865, for the second LMS (μ/10) it was AMSD = 0.20723, for the CA (CoLMS) it was AMSD = 0.02720 and for the CA with modification (9) it was AMSD = 0.02371.5. Simulation resultsThe proposed combined adaptive filter with various types of LMS-based algorithms is implemented for stationary and nonstationary cases in a system identificationsetup.Performance of the combined filter is compared with the individual ones, that compose the particular combination.In all simulations presented here, the reference dk is corrupted by a zero-meanuncorrelated Gaussian noise with 1.02=nσand SNR = 15 dB. Results are obtained by averaging over 100 independent runs, with N = 4, as in the previous section.(a) Time varying optimal weighting vector: The proposed idea may be applied to the SA algorithms in a nonstationary case. In the simulation, the combined filter is composed out of three SA adaptive filters with different steps, i.e. Q = {μ, μ/2, μ/8}; μ = 0.2. The optimal vectors is generated according to the presented model with 001.02=Z σ,and with κ = 2. In the first 30 iterations the variance was estimated according to (7), and CA takes the coefficients of SA with μ (SA1).Figure 2(a) shows the AMSD characteristics for each algorithm. In steady statethe CA does not ideally follow the SA3 with μ/8, because of the nonstationary problem nature and a relatively small difference between the coefficient variances of the SA2 and SA3. However,this does not affect the overall performance of the proposed algorithm. AMSD for each considered algorithm was: AMSD = 0.4129 (SA1,μ), AMSD = 0.4257 (SA2,μ/2), AMSD = 1.6011 (SA3, μ/8) and AMSD = 0.2696 (Comb).(b) Comparison with VS LMS algorithm [6]: In this simulation we take the improved CA (9) from 3.1, and compare its performance with the VS LMS algorithm [6], in the case of abrupt changes of optimal vector. Since the considered VS LMS algorithm[6] updates its step size for each weighting coefficient individually, the comparison of these two algorithms is meaningful.All the parameters for the improved CA are the same as in 3.1. For the VS LMS algorithm [6], the relevant parameter values are the counter of sign change m0 = 11,and the counter of sign continuity m1 = 7. Figure 2(b)shows the AMSD for the compared algorithms, where one can observe the favorable properties of the CA, especially after the abrupt changes. Note that abrupt changes are generated by multiplying all the system coefficients by −1 at the 2000-th iteration (Fig. 2(b)). The AMSD for the VS LMS was AMSD = 0.0425, while its value for the CA (CoLMS) was AMSD = 0.0323.For a complete comparison of these algorithms we consider now their calculation complexity, expressed by the respective increase in number of operations with respect to the LMS algorithm. The CA increases the number of requres operations for N additions and N IF decisions.For the VS LMS algorithm, the respective increase is: 3N multiplications, N additions, and at least 2N IF decisions.These values show the advantage of the CA with respect to the calculation complexity.6. ConclusionCombination of the LMS based algorithms, which results in an adaptive system that takes the favorable properties of these algorithms in tracking parameter variations, is proposed.In the course of adaptation procedure it chooses better algorithms, all the way to the steady state when it takes the algorithm with the smallest variance of theweighting coefficient deviations from the optimal value.Acknowledgement. This work is supported by the Volkswagen Stiftung, Federal Republic of Germany.基于LMS算法的自适应组合滤波器()()k W bias i 是加权系数的偏差,()k i ρ与方差2σ是零均值的随机变量差,它取决于LMS 的算法类型,以及外部噪声方差2n σ。
自适应滤波器翻译作业概要
第八章快速横向LMS滤波算法8.1 简介在大量的算法解决最小二乘问题递归形式的方法中快速横向递归最小二乘(FTRLS算法是非常具有吸引力,因为其能减少计算复杂度。
FTRLS算法可以通过求解同时向前和向后的线性预测问题,连同其他两个横向过滤器:过程估计量和一个辅助滤波器的期望信号向量有一个作为其第一和唯一的非零元素(例如,d(0= 1。
与格型算法相比,FTRLS算法只需要时间递归方程。
然而,需要得到一些FTRLS算法的关系,可参考前面一章LRLS算法。
FTRLS算法考虑快速的横向滤波器RLS的算法更新的解决方法。
因为顺序固定,更新横向自适应滤波器系数向量在每个计算中都迭代。
格型算法的向后和向前的派生关系可以用于预测所派生的FTRLS算法。
由此产生的算法计算复杂度在实际中实现N使它们特别具有吸引力。
相比格型算法,FTRLS算法的计算复杂度较低,由于没有权向量更新方程。
特别是,FTRLS算法通常需要7 n到11 n每输出样本,乘法和除法则需要LRLS 14n到29 n计算。
因此,FTRLS算法被认为是最快的解决方案的实现RLS的问题[1]-[7]。
在工程实践领域相继提出几种不同的FTRLS算法,所谓的快速卡尔曼算法[1],这的确是一个早期的快速横向RLS算法,计算11n次乘法和除法的复杂运算在每次输出示例。
在后面的研究阶段开发领域的快速横向算法,快速后验误差序列的技术(fa[2],快速横向滤波器(FTF[3]算法提出了要求,同样需要7n乘法和每次除法的输出样本。
FTF算法是具有最低的复杂性的RLS算法,不幸的是,这些算法对量子化效应非常敏感,如果有一些步骤没被采取将会变得不稳定。
在这一章,FTRLS算法的一种特殊形式将被提到,基于那些被提的网格算法所派生出来的。
众所周知,量子化错误在FTRLS算法中是指数发散 [1]-[7]。
自从FTRLS算法不稳定的行为用有限精度算法实现的时候,我们讨论实现FTRLS数值稳定的算法,并提供一个特定算法的描述[8],[10]。
lms自适应滤波
自适应滤波原理:
Lms– least mean square
自适应滤波是一套反馈控制系统,是指利用前一时刻的结果,自动调节当前时刻的滤波器参数,以适应信号和噪声未知或随机变化的特性,得到有效的输出,主要由参数可调的数字滤波器和自适应算法两部分组成,如图1所示
图1自适应滤波器原理图
∙首先假设输入信号是所要信号和干扰噪声之和
∙可变滤波器有有限脉冲响应结构,这样结构的脉冲响应等于滤波器系数。
阶滤波器的系数定义为
.
∙误差信号或者叫作代价函数,是所要信号与估计信号之差
可变滤波器通过将输入信号与脉冲响应作卷积估计所要信号,用向量表示
为
其中
是输入信号向量。
另外,可变滤波器每次都会马上改变滤波器系数
其中是滤波器系数的校正因子。
自适应算法根据输入信号与误差信号生成这个校正因子,根据修改矫正因子的方式不同可以区分为不同的自适应算法。
自适应滤波器的自适应过程是:用自适应算法(Update Algorithm)调节FIR或IIR滤波器的系数,使误差信号逼近于0。
疑问:在实际应用中如何获得d(n)期望输出信号?或者如何获得噪声信号?。
LMS自适应滤波器
LMS自适应滤波器是使滤波器的输出信号与期望响应之间的误差的均方值为最小,因此称为最小均方(LMS)自适应滤波器。
function [yn,W,en]=LMS(xn,dn,M,mu,itr)% LMS(Least Mean Squre)算法%输入参数:% xn 输入的信号序列(列向量)% dn 所期望的响应序列 (列向量)% M 滤波器的阶数(标量)% mu 收敛因子(步长) (标量)要求大于0,小于xn的相关矩阵最大特征值的倒数% itr 迭代次数 (标量) 默认为xn的长度,M<itr<length(xn)% 输出参数:% W 滤波器的权值矩阵 (矩阵)% 大小为M x itr,% en 误差序列(itr x 1)(列向量)% yn 实际输出序列(列向量)%参数个数必须为4个或5个if nargin == 4 % 4个时递归迭代的次数为xn的长度itr = length(xn);elseif nargin == 5 % 5个时满足M〈itr〈length(xn)if itr〉length(xn) | itr〈Merror(’迭代次数过大或过小!');endelseerror(’请检查输入参数的个数!’);end%初始化参数en = zeros(itr,1);%误差序列,en(k)表示第k次迭代时预期输出与实际输入的误差W = zeros(M,itr); % 每一行代表一个加权参量,每一列代表-次迭代,初始为0%迭代计算for k = M:itr % 第k次迭代x = xn(k:—1:k-M+1);%滤波器M个抽头的输入 y = W(:,k-1).' * x;%滤波器的输出en(k) = dn(k)— y ; % 第k次迭代的误差% 滤波器权值计算的迭代式W(:,k) = W(:,k—1) + 2*mu*en(k)*x;end% 求最优时滤波器的输出序列yn = inf * ones(size(xn));for k = M:length(xn)x = xn(k:—1:k—M+1);yn(k) = W(:,end)。
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Combined Adaptive Filter with LMS-Based Algorithms´Abstract: A combined adaptive filter is proposed. It consists of parallel LMS-based adaptive FIR filters and an algorithm for choosing the better among them. As a criterion for comparison of the considere d algorithms in the proposed filter, we take the ratio between bias and variance of the weighting coefficients. Simulations results confirm the advantages of the proposed adaptive filter.Keywords: Adaptive filter, LMS algorithm, Combined algorithm,Bias and var iance trade-off1.IntroductionAdaptive filters have been applied in signal processing and control, as well as in many practical problems, [1, 2]. Performance of an adaptive filter depends mainly on the algorithm used for updating the filter weighting coeffici ents. The most commonly used adaptive systems are those based on the Least Mean Square (LMS) adaptive algorithm and its modifications (LMS-based algorithms).The LMS is simple for implementation and robust in a number of applications [1–3]. However, since it does not always converge in an acceptable manner, there have been many attempts to improve its performance by the appropriate modifications: sign algorithm (SA) [8], geometric mean LMS (GLMS) [5], variable step-size LMS(VS LMS) [6, 7].Each of the LMS-bas ed algorithms has at least one parameter that should be defined prior to the adaptation procedure (step for LMS and SA; step and smoothing coefficients for GLMS; various parameters affecting the step for VS LMS). These parameters crucially influence the filter output during two adaptation phases:transient and steady state. Choice of these parameters is mostly based on some kind of trade-off between the quality of algorithm performance in the mentioned adaptation phases. We propose a possible approach for the LMS-based adaptive filter performance improvement. Namely, we make a combination of several LMS-based FIR filters with different parameters, and provide the criterion for choosing the most suitable algorithm for different adaptation phases. This method may be applied to all theLMS-based algorithms, although we here consider only several of them.The paper is organized as follows. An overview of the considered LMS-based algorithms is given in Section 2.Section 3 proposes the criterion for evaluation and combination of adaptive algorithms. Simulation results are presented in Section 4.2. LMS based algorithms Let us define the input signal vector T k N k x k x k x X )]1()1()([+--= and vector of weighting coefficients as T N k k W k W k W W )]()()([110-= .The weighting coefficients vector should be calculated according to:}{21k k k k X e E W W μ+=+ (1) where µ is the algorithm step, E{·} is the estimate of the expected value and k T k k k X W d e -=is the error at the in-stant k,and dk is a reference signal.Depending on the estimation of expected value in (1), one defines various forms of adaptive algorithms: the LMS {}()k k k k X e X e E =,the GLMS {}()()∑=--≤<-=k i i k i k i k k a X e a a X e E 010,1, and the SA {}()()k k k k e sign X X e E =,[1,2,5,8] .The VS LMS has the same form as the LMS, but in the adaptation the step µ(k) is changed [6, 7].The considered adaptive filtering problem consists in trying to adjust a set of weighting coefficients so that the system output,k T k k X W y =, tracks a reference signal, assumed as k k Tk k n X W d +=*,where k n is a zero mean Gaussian noise with thevariance 2n σ,and *k W is the optimal weight vector (Wiener vector). Two cases will be considered:W W k =* is a constant (stationary case) and *k W is time-varying (nonstationary case). In nonstationary case the unknown system parameters( i.e. the optimal vector *k W )are time variant. It is often assumed that variation of *k W may be modeled as K k k Z W W +=+**1 is the zero-mean random perturbation, independent onk X and k n with the autocorrelation matrix []I Z Z E G Z T k k 2σ==.Note that analysisfor the stationary case directly follows for 02=Z σ.The weighting coefficient vector converges to the Wiener one, if the condition from [1, 2] is satisfied.Define the weighting coefficientsmisalignment, [1–3],*k k k W W V -=. It is due to both the effects of gradient noise (weighting coefficients varia tions around the average value) and the weighting vector lag (difference between the average and the optimal value), [3]. It can be expressed as:()()()()*k k k k k W W E W E W V -+-=, (2)According to (2), the ith element of k V is:(3) where ()()k W bias i is the weighting coefficient bias and ()k i ρ is a zero-mean random variable with the variance 2σ.The variance depends on the type ofLMS-based algorithm, as well as on the external noise variance 2n σ.Thus, if the noisevariance is constant or slowly-varying,2σ is time invariant for a particular LMS-based algorithm. In that sense, in the analysis that follows we will assume that 2σ depends only on the algorithm type, i.e. on its parameters.An important performance measure for an adaptive filter is its mean square deviation (MSD) of weighting coefficients. For the adaptive filters, it is given by, [3]:[]k T k k V V E MSD ∞→=lim .3. Combined adaptive filterThe basic idea of the combined adaptive filter lies in parallel implementation of two or more adaptive LMS-based algorithms, with the choice of the best among them in each iteration [9]. Choice of the most appropriate algorithm, in each iteration, reduces to the choice of the best value for the weighting coefficients. The best weighting coefficient is the one that is, at a given instant, the closest to the corresponding value of the Wiener vector.Let ()q k W i , be the i −th weighting coefficient for LMS -based algorithm with the chosen parameter q at an instant k. Note that one may now treat all the algorithms in a unified way (LMS: q ≡ µ,GLMS: q ≡ a,SA:q ≡ µ). LMS -based algorithm behavior is crucially dependent on q. In each iteration there is an optimal value qopt , producing the best performance of the adaptive al-gorithm. Analyze no w a combined adaptive filter, with several LMS -based algorithms of the same type, but with different parameter q.The weighting coefficients are random variables distributed around the ()k W i *,with()()q k W bias i ,and the variance 2q σ, related by [4, 9]: ()()()()q i i i q k W bias k W q k W κσ≤--,,*, (4)()()()()()()()()()()()()k k W bias k W E k W k W k W E k V i i i i i i i ρ+=-+-=*where (4) holds with the probability P(κ), dependent on κ. For example, for κ = 2 and a Gaussian distribution,P(κ) = 0.95 (two sigma rule).Define the confidence intervals for ()]9,4[,,q k W i :()()()[]q i q i i q k W k q k W k D κσσ2,,2,+-= (5) Then, from (4) and (5) we conclude that, as long as ()()q i q k W bias κσ<,,()()k D k W i i ∈*, independently on q. This means that, for small b ias, the confidence intervals, for different s q ' of the same LMS-based algorithm, of the same LMS-based algorithm, intersect. When, on the other hand, the bias becomes large, then the central positions of the intervals for different s q ' are far apart, and they do not intersect.Since we do not have apriori information about the ()()q k W bias i ,,we will use a specific statistical approach to get the criterion for the choice of adaptive algorithm, i.e. for the values of q. The criterion follows from the trade-off condition that bias and variance are of the same order of magnitude, i.e.()()[]4,,q i q k W bias κσ≅.The proposed combined algorithm (CA) can now be summarized in the following steps:Step 1. Calculate ()q k W i ,for the algorithms with different s q 'from the predefined set {} ,,2q q Q i =.Step 2. Estimate the variance 2q σ for each considered algorithm.Step 3. Check if ()k D i intersect for the considered algorithms. Start from an algorithm with largest value of variance, and go toward the ones with smaller values of variances. According to (4), (5) and the trade-off criterion, this check reduces to the check if()()()ql qm l i m i q k W q k W σσκ+<-2,, (6)is satisfied, where Q q q l m ∈,,and the following relation holds:Q q q h ql qh qm h ∉⇒>>∀,:222σσσ.If no ()k D i intersect (large bias) choose the algorithm with largest value of variance. If the ()k D i intersect, the bias is already small. So, check a new pair of weighting coefficients or, if that is the last pair, just choose the algorithm with the smallest variance. First two intervals that do not intersect mean that the proposed trade-off criterion is achieved, and choose the algorithm with large variance.Step 4. Go to the next instant of time.The smallest number of elements of the set Q is L =2. In that case, one of the s q 'should provide good tracking of rapid variations (the largest variance), while the other should provide small variance in the steady state. Observe that by adding few more s q ' between these two extremes, one may slightly improve the transient behavior of the algorithm.Note that the only unknown values in (6) are the variances. In our simulations weestimate 2q σ as in [4]:()()()2675.0/1--=k W k W mediani i q σ, (7) for k = 1, 2,... , L and 22qZ σσ<<. The alternative way is to estimate 2n σ as:∑=≈T i i n e T 1221σ,for x(i) = 0. (8) Expressions relating 2n σ and 2q σ in steady state, for different types of LMS-basedalgorithms, are known from literature. For the standard LMS algorithm in steady state,2n σ and 2q σ are related 22nq q σσ=,[3]. Note that any other estimation of 2q σis valid for the proposed filter.Complexity of the CA depends on the constituent algorithms (Step 1), and on the decision algorithm (Step 3).Calculation of weighting coefficients for parallel algorithms does not increase the calculation time, since it is performed by a parallel hardware realization, thus increasing the hardware requirements. The variance estimations (Step 2), negligibly contribute to the increase of algorithm complexity, because they are performed at the very beginning of adaptation and they are using separate hardware realizations. Simple analysis shows that the CA increases the number of operations for, at most, N(L −1) additions and N(L −1) IF decisions, and needs some additional hardware with respect to the constituent algorithms.4.Illustration of combined adaptive filterConsider a system identification by the combination of two LMS algorithms with different steps. Here, the parameter q is μ,i.e. {}{}10/,,21μμ==q q Q .The unknown system has four time-invariant coefficients,and the FIR filters are with N = 4. We give the average mean square deviation (AMSD ) for both individual algorithms, as well as for their combination,Fig. 1(a). Results are obtained byaveraging over 100 independent runs (the Monte Carlo method), with μ = 0.1. Thereference dk is corrupted by a zero-mean uncorrelated Gaussian noise with 2n σ= 0.01and SNR = 15 dB, and κ is 1.75. In the first 30 iterations the variance was estimated according to (7), and the CA picked the weighting coefficients calculated by the LMS with μ.As presented in Fig. 1(a), the CA first uses the LMS with μ and then, in the steady state, the LMS with μ/10. Note the region, between the 200th and 400th iteration,where the algorithm can take the LMS with either stepsize,in different realizations. Here, performance of the CA would be improved by increasing the number of parallel LMS algorithms with steps between these two extrems.Observe also that, in steady state, the CA does not ideally pick up the LMS with smaller step. The reason is in the statistical nature of the approach.Combined adaptive filter achieves even better performance if the individual algorithms, instead of starting an iteration with the coefficient values taken from their previous iteration, take the ones chosen by the CA. Namely, if the CA chooses, in the k -th iteration, the weighting coefficient vector P W ,theneach individual algorithm calculates its weighting coefficients in the (k +1)-th iteration according to:{}k k p k X e E W W μ21+=+(9)Fig. 1. Average MSD for considered algorithms.Fig. 2. Average MSD for considered algorithms.Fig. 1(b) shows this improvement, applied on the previous example. In order to clearly compare the obtained results,for each simulation we calculated the AMSD . For the first LMS (μ) it was AMSD = 0.02865, for the second LMS (μ/10) it was AMSD = 0.20723, for the CA (CoLMS) it was AMSD = 0.02720 and for the CA with modification (9) it was AMSD = 0.02371.5. Simulation resultsThe proposed combined adaptive filter with various types of LMS-based algorithms is implemented for stationary and nonstationary cases in a system identificationsetup.Performance of the combined filter is compared with the individual ones, that compose the particular combination.In all simulations presented here, the reference dk is corrupted by a zero-meanuncorrelated Gaussian noise with 1.02=n σand SNR = 15 dB. Results are obtained byaveraging over 100 independent runs, with N = 4, as in the previous section.(a) Time varying optimal weighting vector: The proposed idea may be applied to the SA algorithms in a nonstationary case. In the simulation, the combined filter is composed out of three SA adaptive filters with different steps, i.e. Q = {μ, μ/2, μ/8}; μ = 0.2. The optimal vectors is generated according to the presented model with 001.02=Z σ,and with κ = 2. In the first 30 iterations the variance was estimated according to (7), and CA takes the coefficients of SA with μ (SA1).Figure 2(a) shows the AMSD characteristics for each algorithm. In steady statethe CA does not ideally follow the SA3 with μ/8, because of the nonstationary problem nature and a relatively small difference between the coefficient variances of the SA2 and SA3. However,this does not affect the overall performance of the proposed algorithm. AMSD for each considered algorithm was: AMSD = 0.4129 (SA1,μ), AMSD = 0.4257 (SA2,μ/2), AMSD = 1.6011 (SA3, μ/8) and AMSD = 0.2696 (Comb).(b) Comparison with VS LMS algorithm [6]: In this simulation we take the improved CA (9) from 3.1, and compare its performance with the VS LMS algorithm [6], in the case of abrupt changes of optimal vector. Since the considered VS LMS algorithm[6] updates its step size for each weighting coefficient individually, the comparison of these two algorithms is meaningful.All the parameters for the improved CA are the same as in 3.1. For the VS LMS algorithm [6], the relevant parameter values are the counter of sign change m0 = 11,and the counter of sign continuity m1 = 7. Figure 2(b)shows the AMSD for the compared algorithms, where one can observe the favorable properties of the CA, especially after the abrupt changes. Note that abrupt changes are generated by multiplying all the system coefficients by −1 at the 2000-th iteration (Fig. 2(b)). The AMSD for the VS LMS was AMSD = 0.0425, while its value for the CA (CoLMS) was AMSD = 0.0323.For a complete comparison of these algorithms we consider now their calculation complexity, expressed by the respective increase in number of operations with respect to the LMS algorithm. The CA increases the number of requres operations for N additions and N IF decisions.For the VS LMS algorithm, the respective increase is: 3N multiplications, N additions, and at least 2N IF decisions.These values show the advantage of the CA with respect to the calculation complexity.6. ConclusionCombination of the LMS based algorithms, which results in an adaptive system that takes the favorable properties of these algorithms in tracking parameter variations, is proposed.In the course of adaptation procedure it chooses better algorithms, all the way to the steady state when it takes the algorithm with the smallest variance of theweighting coefficient deviations from the optimal value.Acknowledgement. This work is supported by the Volkswagen Stiftung, Federal Republic of Germany.基于LMS算法的自适应组合滤波器摘要:提出了一种自适应组合滤波器。