循环神经网络自适应有源噪声对消研究

介绍了一种新颖的非线性逢适应滤波器－－循环神经网络自适应滤波器。由于这种循环神经等效于非线性ＩＩＲ滤波器,具有学习非线性函数到任意的精度及自适应能力,这种滤波器优于线性滤波器,能够适应各种噪声环境。本文将该滤波器用于有源噪声对消,仿真结果表明了这种循环神经网络自适应有源噪声对消系统具有良好的抗噪声性能。     

基于sigmoid函数的Volterra自适应有源噪声对消器

该文介绍了一种新颖的非线性自适应有源噪声对消器基于sigmoid函数的Volterra自适应有源噪声对消器,并采用输入信号和瞬时误差归一化的LMS自适应算法调整其系数。这种基于sigmoid函数的Volterra自适应有源噪声对消器具有参数少和便于实现的模快化结构等优点。仿真结果表明:这种基于sigmoid函数的Volterra自适应有源噪声对消系统具有良好的抗噪声性能。     

基于Lyapunov稳定性理论的Volterra自适应噪声对消器

介绍了在理稳定分布环境下一种新的非线性Voherra自适应噪声对消器。由于Ⅱ稳定分布噪声有显著的尖峰脉冲特性,Vohem级数的非线性项将其更加放大,严重影响了收敛性能。提出了Voherra自适应噪声对消器利用sigmoid函数对输入α噪声进行非线性预处理,抑制尖峰脉冲的影响,基于Lyapunov稳定性,定义新的Lyapunov函数．给出了二阶Voherra自适应滤波器算法。仿真实验表明,该算法在不同特征指数的稳定分布噪声环境中都表现出了良好的抗噪声性能。     

基于改进Hopfield神经网络的自适应二维噪声对消器

提出了基于连续型Hopfield神经网络(CHNN)的自适应二维噪声对消器,讨论了神经网络的结构和原理及相应的自适应滤波算法,并从理论上进行了论证.仿真结果表明相对于采用最小均方算法的二维线性噪声对消器,CHNN噪声对消器能更有效实现二维噪声的消除,保持原信号的完整性,获得较好的去噪声效果.     

一种基于格型正交化的二阶Volterra自适应滤波算法

针对 Volterra 自适应滤波器输入信号相关性或附加的非线性畸变的增强使自适应滤波器性能下降的问题,本文提出基于格型正交化的二阶 Volterra 自适应滤波算法.先对输入信号进行格型预处理,得到互相正交的后向预测误差信号;然后将其作为自适应滤波器的输入,从而大大降低了一次项、平方项和交叉乘积项信号各项之间的耦合,改善了自适应算法的收敛性能.有源噪声对消的仿真结果表明,在输入噪声强相关和附加较强非线性畸变时本算法仍具有较好的消噪性能.     

一种改进的Volterra自适应噪声对消器

针对随着Voherra滤波器的输入维数或记忆单元增大,相应需要的计算复杂性成幂级数快速增加的问题,提出了一种改进的利用sigmoid函数对背景噪声进行预处理的减少参数Voherra自适应滤波器,并用它构造了非线性自适应噪声对消器。仿真结果表明,算法在高斯噪声和α稳定分布噪声环境下都有优越的抗噪性能。     

基于离散余弦变换的旁瓣对消技术研究

自适应旁瓣对消是一种有效抑制有源干扰的措施。研究了自适应旁瓣对消和合成孔径雷达（SAR）有源遮盖式干扰的基本原理,详细推导了基于离散余弦变换的DCT-LMS频域自适应方法,并将其应用于SAR的旁瓣对消系统中。通过与其他自适应算法的对比实验,证明了DCT-LMS算法兼有收敛速度快,计算量小的优点。最终模拟实际环境中的干扰源,利用SAR的实际数据进行了仿真实验。实验结果表明,DCT-LMS算法能有效地抑制有源干扰噪声,确保SAR接收机正常工作,具有较高的干扰对消比。     

噪声对消在信号处理系统中的应用

论述基于LMS算法的自适应滤波器噪声对消的工作原理,以及基于AR模型的信号分析方法。在这两种方法相结合的情况下,能有效去除信号的噪声。对含有瞬态干扰的微弱信号,用AR模型法估计出噪声的系数并预测噪声波形,通过自适应滤波器噪声对消原理进行滤波,最后在Matlab环境下进行仿真试验,结果表明该方法具有较好的去噪效果。     

主从结构自适应噪声对消器MSANC的最优步长函数选择

ＭＳＡＮＣ是一种新型优化的主从结构自适应声对消器,此系统具有两个自适应滤波器,主滤波器和从滤波器。从滤波器用于估计输入信噪比,主滤波器进行真正的自适应滤波。主滤波器的步长是输入信噪比的函数,对这个步长函数的选取,我们根据函数类型考虑了几种不情况,正指数型函数能取得最佳对消效果,是针对于此系统的最优选择。     

抖动偏频激光陀螺抖动信号的自适应对消

根据抖动偏频激光陀螺的输入、输出信号模型,提出利用自适应噪声对消技术消除陀螺抖动信号.使用恰当的算法,自适应噪声对消器能够自适应地跟踪并最佳地消除抖动信号,提取所需的惯性角速率信息.     

IP电话回声消除器中自适应滤波器的研究

介绍了一种用于IP电话的自适应回声消除器，采用LMS自适应滤波器实现。首先通过Matlab进行LMS算法的步长选择，并用生成的符合G．168要求的带限CSS测试信号对回声消除器的性能进行了测试。     

Nonlinear adaptive bilinear filters for active noise control systems

The reference and error channels of active noise control (ANC) systems may be saturated in real-world applications if the noise level exceeds the dynamic range of the electronic devices. This nonlinear saturation degrades the performance of ANC systems that use linear adaptive filters with the filtered-X least-mean-square (FXLMS) algorithm. This paper derives a bilinear FXLMS algorithm for nonlinear adaptive filters to solve the problems of signal saturation and other nonlinear distortions that occur in ANC systems used for practical applications. The performance of this bilinear adaptive filter is evaluated in terms of convergence speed, residual noise in steady state, and the computational complexity for different filter lengths. Computer simulations verify that the nonlinear adaptive filter with the associated bilinear FXLMS algorithm is more effective in reducing saturation effects in ANC systems than a linear filter and a nonlinear Volterra filter with the FXLMS algorithm.     

A minimum misadjustment adaptive FIR filter

The performance of an adaptive filter is limited by the misadjustment resulting from the variance of adapting parameters. This paper develops a method to reduce the misadjustment when the additive noise in the desired signal is correlated. Given a static linear model, the linear estimator that can achieve the minimum parameter variance estimate is known as the best linear unbiased estimator (BLUE). Starting from classical estimation theory and a Gaussian autoregressive (AR) noise model, a maximum likelihood (ML) estimator that jointly estimates the filter parameters and the noise statistics is established. The estimator is shown to approach the best linear unbiased estimator asymptotically. The proposed adaptive filtering method follows by modifying the commonly used mean-square error (MSE) criterion in accordance with the ML cost function. The new configuration consists of two adaptive components: a modeling filter and a noise whitening filter. Convergence study reveals that there is only one minimum in the error surface, and global convergence is guaranteed. Analysis of the adaptive system when optimized by LMS or RLS is made, together with the tracking capability investigation. The proposed adaptive method performs significantly better than a usual adaptive filter with correlated additive noise and tracks a time-varying system more effectively     

回声消除器中自适应滤波器的研究

文中介绍了一种用于IP电话的自适应回声消除器,采用NLMS算法自适应滤波器来实现.首先通过MATLAB进行了NLMS算法的步长选择,并用生成的符合G.168要求的带限CSS测试信号对回声消除器的性能进行了测试.     

物联传感网噪声过滤的自适应线性权值算法

针对物联传感网络由于数据类型不一致造成散粒噪声.提出一种物联传感网络散粒噪声过滤的自适应线性权值算法.提取物联传感网络散粒噪声特征参数,对上述参数进行线性变换,获取噪声关联特征参数自适应线性权值,完成物联传感网络散粒噪声过滤.实验表明,该算法能够有效过滤物联传感网络中的散粒噪声,取得了令人满意的结果.     

Robust adaptive recovery of spread-spectrum signals with short data records

The problem under consideration is the adaptive reception of a multipath direct-sequence spread-spectrum (SS) signal in the presence of unknown correlated SS interference and additive impulsive noise. An SS receiver structure is proposed that consists of a vector of adaptive chip-based Hampel nonlinearities followed by an adaptive auxiliary-vector linear tap-weight filter. The nonlinear receiver front end adapts itself to the unknown prevailing noise environment providing robust performance over a wide range of underlying noise distributions. The adaptive auxiliary-vector linear tap-weight filter allows rapid SS interference suppression with a limited data record. Numerical and simulation studies under finite-data-record system adaptation show significant improvement in bit-error-rate performance over the conventional linear minimum variance-distortionless-response (MVDR) SS receiver or conventional MVDR filtering preceded by vector adaptive chip-based nonlinear processing.     

噪声有源控制的人工神经网络方法

讨论了有源噪声控制（ＡＮＣ）问题,提出一种基于人工神经网络的非线性噪声有源自适应控制方法,给出了一种基于误差梯度下降的学习算法,证明了闭环控制系统在Ｌｙａｐｕｎｏｖ意义下的稳定性。     

A new adaptive recursive RLS-based fast-array IIR filter for active noise and vibration control systems

Infinite impulse response filters have not been used extensively in active noise and vibration control applications. The problems are mainly due to the multimodal error surface and instability of adaptive IIR filters used in such applications. Considering these, in this paper a new adaptive recursive RLS-based fast-array IIR filter for active noise and vibration control applications is proposed. At first an RLS-based adaptive IIR filter with computational complexity of order O(n²) is derived, and a sufficient condition for its stability is proposed by applying passivity theorem on the equivalent feedback representation of this adaptive algorithm. In the second step, to reduce the computational complexity of the algorithm to the order of O(n) as well as to improve its numerical stability, a fast array implementation of this adaptive IIR filter is derived. This is accomplished by extending the existing results of fast-array implementation of adaptive FIR filters to adaptive IIR filters. Comparison of the performance of the fast-array algorithm with that of Erikson’s FuLMS and SHARF algorithms confirms that the proposed algorithm has faster convergence rate and ability to reach a lower minimum mean square error which is of great importance in active noise and vibration control applications.