基于谱减和LMS的自适应语音增强

针对宽带噪声背景下的语音增强问题,将短时语音视为非平稳或宽平稳信号,基于谱减法和自适应滤波的最小均方（LMS）算法,提出了一种FIR型自适应滤波算法（SSLMS）：用减谱法由短时噪声观测语音估计期望信号,作为滤波器输出信号的参考信号;用滤波器的输出与参考信号的差值为误差信号,用LMS算法求得滤波器权系数修正量,并修正滤波器。权系数最速下降调整中,采用了归一化LMS、符号LMS、块LMS技术,以简化保证权系数收敛的步长选择、减少权系数修正的运算量,从而提高自适应速度。对不同的语音在各种信噪比下仿真实验,并与改进的谱减法比较,结果表明,该法增强效果优于谱减法;在信噪比为3 dB时该法的增强效果仍然令人满意。     

基于LMS算法自适应噪声抵消器的分析研究

自适应信号处理的理论和技术已经成为人们常用的语音去噪技术,而Matlab为其提供了更为方便快捷的方法来对语音信号进行消噪处理。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,重点研究了LMS自适应滤波算法,并对LMS自适应算法进行了分析,用Matlab对其进行了仿真和实现。     

A new adaptive center weighted median filter for suppressing impulsive noise in images

In this work, a new adaptive center weighted median (ACWM) filter is proposed for improving the performance of median-based filters, preserving image details while effectively suppressing impulsive noise. The proposed filter is an adaptive CWM filter with an adjustable central weight obtained by partitioning the observation vector space. To obtain the optimal weight for each block, the efficient scalar quantization (SQ) method is used to partition the observation vector space. The center weight within each block is obtained by using a learning approach based on the least mean square (LMS) algorithm. The noise filtering procedure is progressively applied through several iterations so that the mean square error of the output can be minimized. Experimental results have demonstrated that the proposed filter outperforms many well-accepted median-based filters in terms of both noise suppression and detail preservation. The proposed new filter also provides excellent robustness at various percentages of impulsive noise.     

一种新变步长LMS算法及在自适应波束形成中的应用

LMS算法是智能天线自适应波束形成算法中的经典算法,由于其步长固定,造成收敛速度和稳态失调之间的矛盾。为了解决这一问题,提出一种新的变步长LMS算法,并在算法中引入误差信号的自相关估计,大大降低了噪声的干扰,对算法进行仿真,得到了最优的参数设置。利用蒙特卡罗方法对算法进行了性能评估,与传统的LMS算法、NLMS算法相比,新的变步长LMS算法具有更快的收敛速度和较小的稳态误差及优良的抗噪性能。     

自适应滤波器在电参数测量系统中的应用

论述了基于LMS算法的自适应噪声抵消器和自适应谱线增强器的工作原理及改进的算法,并将自适应噪声抵消器和自适应谱线增强器合并为一个滤波器应用于电参数测量系统中,仿真实验证明这种滤波器去除噪声的性能较佳.     

Partition-based fuzzy median filter based on adaptive resonance theory

This paper presents a novel partition-based fuzzy median filter for noise removal from corrupted digital images. The proposed filter is obtained as the weighted sum of the current pixel value and the output of the median filter, where the weight is set by using fuzzy rules concerning the state of the input signal sequence to indicate to what extent the pixel is considered to be noise. Based on the adaptive resonance theory, the authors developed a neural network model and created a new weight function where the neural network model is employed to partition the observation vector. In this framework, each observation vector is mapped to one of the M blocks that form the observation vector space. The least mean square (LMS) algorithm is applied to obtain the optimal weight for each block. Experiment results have confirmed the high performance of the proposed filter in efficiently removing impulsive noise and Gaussian noise.     

一种新的LMS自适应滤波算法分析仿真研究

传统变步长最小均方(LMS)算法存在收敛速度慢、易受噪声干扰等缺点,为了提高算法的性能,通过对变步长LMS算法进行分析研究,在步长因子x(n)与误差信号e(n)的相关统计量之间建立一种新的非线性函数关系,提出了一种新的变步长LMS自适应滤波算法。该算法采用误差信号的自相关时间均值来调节步长,并用绝对估计误差的扰动量以加快自适应滤波器抽头权向量的收敛。理论分析与计算机仿真结果表明:与SVSLMS和G-SVSLMS算法比较,该算法具有较快的收敛速度、较小的稳态误差以及较强的抗干扰能力。     

一种复合式宽带有源噪声控制算法

为获得更高的宽带噪声控制效果,提出了一种复合结构有源噪声控制算法.该算法将传统FXLMS算法和DFT-FSF算法并行运行,实现对宽带和窄带噪声的同时降噪.新的复合式算法对正弦波噪声实现高达45dB的降噪,而对带限白噪声的平均降噪量则达到15dB.仿真结果证明了算法的有效性.     

一种基于算子理论的去相关LMS自适应滤波算法

针对输入信号向量序列之间的相关性将显著降低LMS算法的性能这一问题,从算子的角度出发,提出了一种新的去相关LMS自适应滤波算法。通过将最新输入向量向以前所有时刻的输入向量序列所张成的线性空间的零空间作正交投影,达到提取新信息的目的,并以提取的新息作为LMS算法的更新方向向量。仿真分析表明,新算法具有收敛速度快、输出误差小以及对信噪比不敏感等特点,并且采用较低的滤波器阶数即可得到良好的滤波效果,同时提高算法的运算效率。     

Computational noise effects on adaptive filter algorithms

Finite word length arithmetic roundoff noise in adaptive filter algorithms results in statistical variations in the filter weight vector about the infinite precision arithmetic weight vector. These roundoff errors may be modeled as a statistically non stationary driving noise affecting weight mean and covariance convergence. Mean and covariance expressions and bounds are desired for word lengths in fixed-point arithmetic by making use of multiplication roundoff error models. The adaptive filter algorithms consist of the LMS algorithm, the Widrow-Hoff LMS algorithm, pilot-vector algorithm and clipped vector algorithm. All of these algorithms can be implemented on-line and real-time. However, only the behavior of the LMS algorithm is reported here. The implementation of the adaptive filter algorithms in finite word length arithmetic is most evident in minicomputer, microprocessor, and dedicated digital signal processors for on-line real-time signal identification and parameter estimation in many disciplines. Radar signal processing, adaptive beam forming, acoustic signal identification, communication channel enhancement have a definite need for advanced filtering concepts. Our adaptive algorithms are typically employed in these filter configurations. These filters can also be employed in phase distortion equalizers. A particular advantage of these filters is that they can be trained to equalize a variety of distortions. Should a particular distortion scenario change in time, the filters can be made to easily adapt to the new problem.     

Partition belief median filter based on Dempster–Shafer theory for image processing

A novel median-type filter controlled by evidence fusion is proposed for removing noise from images. The fusion of evidence based on the Dempster–Shafer evidence theory, providing a way to deal with the uncertainty in the evidence fusion, indicates to what extent a noise is considered. The filter proposed here is obtained as a weighted sum of the current pixel value and the output of the median filter, and the weight is set based on the belief value of the input signal sequence. The efficient step-like function is used to partition the belief space, and the least mean square (LMS) algorithm is applied to obtain the optimal weight for each block. Moreover, to improve the performance, the new filter is recursively implemented. Experimental results have demonstrated that the proposed filter can outperform many well-accepted median-based filters in preserving image details while effectively suppressing impulsive noises, and it also works satisfactorily in reducing Gaussian as well as the mixture of Gaussian and impulsive noise.     

基于声门脉冲的自适应时延估计算法*

在研究LMS自适应算法的基础上,提出一种基于声门脉冲的变步长LMS自适应时延估计新方法,并在相关噪声和混响的环境下与互功率谱相位广义互相关法(GCC-CSP)、变步长LMS自适应算法进行性能比较.实验结果表明,新方法具有很好的鲁棒性,即使在低信噪比强混响的环境下也能获得有效的时延估计.     

一种噪声抑制的线性共轭多用户检测算法

当权向量受到噪声的影响时,最小输出能量（MOE）检测器的性能将显著下降．针对这一问题,设计了一种噪声抑制的线性共轭MOE检测器．将约束最小均方（LMS）算法应用到新的MOE检测器,提出一种基于约束LMS的盲噪声抑制线性共轭MOE多用户检测算法．该算法消除了权向量中的噪声分量,利用了接收向量的复数共轭,从而提高了系统的输出信干噪比和误码率性能．仿真结果表明所提算法有较好的性能．     

免疫算法自适应滤波器对白噪声的抑制

实际信号经常会受到白噪声及高次谐波的影响,由于白噪声频谱分布在整个实数域,常用的滤波器很难将其滤除。讲述了自适应滤波器的原理及用免疫算法自适应滤波器,对白噪声及高次谐波进行抑制的方法。通过免疫算法对自适应滤波器的权向量进行优化,并用均值滤波的方法对自适应滤波器的滤波结果进行进一步滤波,然后用MATLAB对该算法进行仿真。将免疫算法自适应滤波器的仿真结果和LMS滤波算法的仿真结果进行比较,表明免疫算法自适应滤波器能对白噪声及高次谐波进行有效的抑制。     

改进的最小均方自适应滤波算法

针对传统的固定步长最小均方(LMS)算法应用于雷达杂波自适应滤波器系统存在收敛速度与收敛精确度相矛盾的问题,提出一种新的变步长LMS自适应滤波算法。在其基础步长迭代公式中,通过组合自相关误差与前一步长因子来实时更新迭代下一步长因子的方法,达到具有较快的收敛速度和较小的失调,并且不受已经存在的不相关噪声的干扰的效果。仿真结果表明,所提方法的实验效果与传统固定步长LMS算法及已有算法相比,在收敛速率、收敛精度、抑制噪声方面都有很大的改善,证明所提算法是有效、可行的,且与理论分析一致。     

An iterative longest matching segment approach to speech enhancement with additive noise and channel distortion

This paper presents a new approach to speech enhancement from single-channel measurements involving both noise and channel distortion (i.e., convolutional noise), and demonstrates its applications for robust speech recognition and for improving noisy speech quality. The approach is based on finding longest matching segments (LMS) from a corpus of clean, wideband speech. The approach adds three novel developments to our previous LMS research. First, we address the problem of channel distortion as well as additive noise. Second, we present an improved method for modeling noise for speech estimation. Third, we present an iterative algorithm which updates the noise and channel estimates of the corpus data model. In experiments using speech recognition as a test with the Aurora 4 database, the use of our enhancement approach as a preprocessor for feature extraction significantly improved the performance of a baseline recognition system. In another comparison against conventional enhancement algorithms, both the PESQ and the segmental SNR ratings of the LMS algorithm were superior to the other methods for noisy speech enhancement.     

基于自适应卡尔曼滤波的盲多用户检测算法

对强多址干扰情况下码分多址系统的盲多用户检测算法进行研究,针对多径信道的码分多址系统,提出一种基于自适应卡尔曼滤波的盲多用户检测算法。该算法可在进行状态滤波的同时对未知噪声的统计特性进行在线估计,确保算法收敛于期望用户,提高检测器在动态环境下的跟踪性能。仿真结果表明,与最小均方算法及递推最小二乘算法相比,该算法具有更好的收敛性和动态性能。     

激光陀螺信号解调中自适应滤波器的设计

在激光陀螺信号解调领域中,在满足高精度的前提下如何降低滤波器的延迟一直是相关院所的研究重点。针对此问题,研究了一种新的激光陀螺滤波处理的方法。这种方法采用LMS自适应滤波器原理,分别把机械抖动抖反馈信号作为滤波器的基本输入,把机抖信号、随机噪声和白噪声作为滤波器的参考信号,然后通过FPGA进行数字滤波以及外围控制,最后给出了滤波器的算法实现以及硬件框图。实验结果表明,LMS自适应滤波器有很好的解调效果,经过滤波后的计数值差值在±1个数以内,且延时为1 ms。     

On the convergence rate performance of the normalizedleast-mean-square adaptation

This paper compares the convergence rate performance of the normalized least-mean-square (NLMS) algorithm to that of the standard least-mean-square (LMS) algorithm, which is based on a well-known interpretation of the NLMS algorithm as a form of the LMS via input normalization. With this interpretation, the analysis is considerably simplified and the difference in rate of parameter convergence can be compared directly by evaluating both the condition number of the normalized and unnormalized input correlation matrix. This paper derives the condition number expressions for the normalized input correlation matrix of which the arbitrary-length filter model is linear with respect to its adaptable parameters and contain only two distinct unnormalized eigenvalues. These expressions, which require that the input samples be statistically stationary and zero-mean Gaussian distributed, provide an important insight into the relative convergence performance of the NLMS algorithm to that of the LMS as a function of filter length. This paper also provides a conjecture which set bounds on the NLMS condition number for any arbitrary number of distinct unnormalized eigenvalues.     

易于硬件实现滤除ECG信号运动干扰的变步长LMS算法

针对移动心电(ECG)信号监测系统中运动干扰难以滤除的问题,提出了一种易于硬件实现的数字自适应变步长最小均方(LMS)算法.通过简化步长因子与输入信号的关系,减少了权值更新系统的运算量;分析传统LMS算法收敛性不稳定的问题,结合迭代次数优化步长因子,提高了算法的收敛性能.对比传统LMS算法,所提算法在运算量增加微小的情况下,收敛性能大幅提升,信噪比(SNR)增加大于14dB.仿真结果表明:算法在心电信号进行实时硬件集成滤除运动干扰方面具有运算量小,滤波效果好等优点.