A new fast Newton‐type adaptive filtering algorithm for stereophonic acoustic echo cancellation (SAEC)

This paper addresses the problem of acoustic echo cancellation. We propose a new version of the fast Newton transversal filter algorithm for stereophonic acoustic echo cancellation applications. This algorithm can be viewed as an efficient implementation of the extended two‐channel fast transversal filter algorithm. Moreover, it fits naturally within the frame of the fast version of the recursive least‐squares (RLS) algorithm, applied to the two‐channel case. To stabilize the proposed two‐channel algorithm, we have adapted and then applied a new numerical stabilization technique that has been proposed recently. The computational complexity of the proposed two‐channel algorithm is less than half the complexity of the fastest two‐channel RLS versions and very close to that of the two‐channel normalized least mean squares algorithm when its predicting part length is chosen to be small. Simulation results and comparisons in term of complexities, convergence speed and tracking with the two‐channel algorithms are presented. Copyright © 2009 John Wiley & Sons, Ltd.     

基于改进Kalmus滤波的悬停无人机检测技术

针对复杂杂波环境下悬停无人机检测问题，提出了一种改进的Kalmus滤波 剩余回波时域均值相消 自适应CFAR联合处理算法，对无人机微多普勒检测，实现空管监视目的。通过改进的Kalmus滤波器进行频域滤波，同时对目标回波高频信号和零频信号抑制，并提高零频附近微多普勒信号增益。采用剩余回波均值相消进行二次滤波，提高无人机高速旋翼的多普勒特征信号信噪比，采用短时傅里叶算法检测目标区域多普勒变化，最后通过恒虚警处理，进一步抑制杂波，提取微多普勒信息。试验结果表明本文算法可以对悬停无人机的旋翼多普勒特征进行有效检测，目标多普勒信号幅值提升了约20 dB，实现低空监视管控目的。     

基于自适应陷波滤波器的伺服系统谐振频率估计及抑制

在使用陷波滤波器抑制伺服系统机械谐振时,需要获取准确的机械谐振频率。为了快速检测出谐振频率,提出了一种基于自适应陷波滤波器(ANF)的机械谐振频率估计算法,通过速度误差信号分析,实现谐振频率在线快速辨识。首先,建立柔性连接伺服系统模型;然后,对ANF频率估计算法进行分析,并且与常用的快速傅里叶变换(FFT)频率检测算法的分析精度和计算速度进行对比。数值比较和仿真验证表明,ANF频率估计算法可以更快地实现谐振频率的精确检测。最后搭建试验平台,以ANF频率估计的结果作为陷波器的中心频率,成功实现了电机转速振荡的抑制,验证了该方法的有效性。     

Steiglitz–McBride adaptive notch filter based on a variable‐step‐size LMS algorithm and its application to active noise control

This paper proposes a new Steiglitz–McBride (SM) adaptive notch filter (SM‐ANF) based on a robust variable‐step‐size least‐mean‐square algorithm and its application to active noise control (ANC). The proposed SM‐ANF not only has fast convergence but also has small misadjustment. The variable‐step‐size algorithm uses the sum of the squared cross correlation between the error signal and the delayed inputs corresponding to the adaptive weights. The cross correlation provides robustness to the broadband signal, which plays the role of noise. The proposed SM‐ANF is computationally simpler than the existing Newton/recursive least‐squares‐type ANF. The frequency response of the new SM‐ANF has a notch depth of about ?25 dB (for each of the three frequencies considered) and has spectral flatness within 5 dB (peak to peak). This robust notch filter algorithm is used as an observation noise canceller for the secondary path estimation of an ANC system based on the SM method. The ANC with proposed SM‐ANF provides not only faster convergence but also an 11‐dB improvement in noise attenuation over the SM‐based ANC without such a SM‐ANF. Copyright © 2015 John Wiley & Sons, Ltd.     

Selective partial update and set‐membership improved proportionate normalized subband adaptive filter algorithms

In this article, the concept of proportionate adaptation is extended to the selective partial update (SPU) and set‐membership (SM) normalized subband adaptive filters (NSAFs), and three proportionate normalized subband adaptive filter algorithms are established. The proposed algorithms are the improved proportionate NSAF (IPNSAF), the SPU improved proportionate NSAF (SPU‐IPNSAF), and the SM‐IPNSAF which are suitable for sparse system identification. When the impulse response of the echo path is sparse, the IPNSAF algorithm has faster convergence than NSAF. The performance of IPNSAF is also suitable for dispersive impulse responses. In SPU‐IPNSAF, the filter coefficients are partially updated rather than the entire filter at every adaptation which reduces the computational complexity of IPNSAF. The SM‐IPNSAF exhibits good performance with significant reduction in the overall computational complexity compared with the ordinary IPNSAF. The simulation results show good performance of the proposed algorithms. Copyright © 2010 John Wiley & Sons, Ltd.     

并联式多级回声抵消器设计研究

在回声抵消研究中,LMS算法由于具有算法简单、易于实现的特点而得到广泛应用.然而,采用LMS算法的自适应回声抵消器的性能仍然有很大的改进余地,可以通过对算法本身的改进和对利用该算法的自适应滤波器的结构的改进达到更好的性能,主要研究了利用LMS算法的AEC自适应滤波器结构上的改进,提出了一种新的滤波器设计方法,该方法通过将采样信号在输入端拆分,分别输入到多个并联的子滤波器中分别处理,得到对回声信号的估计后将它从掺杂有回声信号的接收信号中消除,并在输出端重新组合,成为对原始信号的恢复.这种并联多通道结构可以有效提高回声抵消器的收敛速度、降低运算量.由于这种方法利用了多通道并联自适应滤波器的结构,因此要求非常严格的数据同步和时间控制.仿真结果表明,这种方法对回声抵消具有良好的效果.     

Magnetometer calibration improvement using wavelet and genetic algorithm

Three‐axis magnetometer error and measurement noise influence the accuracy of magnetic measurements. Genetic algorithm (GA) is proposed to calibrate the magnetometer error, and wavelet is proposed for noise cancellation. The noise of a Mag3300 magnetometer and a DM magnetometer were tested within a horizontal barrel shield equipment. Five kinds of wavelet analysis and two kinds of wavelet package were used for noise cancellation, and the performance of different wavelets was compared. Noise of the Mag3300 magnetometer and DM magnetometer were suppressed from 29.6 and 2.3466 to 3.7 and 1.0789 nT, respectively. The scalar error of the Mag3300 magnetometer was tested using a two‐dimensional nonmagnetic rotation equipment and a GSM‐19T proton magnetometer. Scalar calibration performances of the unscented Kalman filter (UKF), the two‐step algorithm, and GA were compared. Experimental results show that GA provides less error intensity (about 370 and 70 nT) than UKF and the two‐step algorithm. In addition, the influence of wavelet on scalar calibration using UKF, the two‐step algorithm, and GA was analyzed. Results show that wavelet improved the scalar calibration performance. Mean error of the Mag3300 magnetometer scalar was 795.5 nT. When combined with the wavelet package, the error was suppressed to −22.3, 3.3, and −0.4 nT using UKF, the two‐step algorithm, and GA, respectively. The results suggest an effective way for magnetometer calibration using GA and wavelet. © 2016 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

基于改进型自适应算法的谐波检测及其性能研究

通过对有源电力滤波器基本自适应谐波电流对消检测方法的研究,指出了基本自适应方法存在检测精度和动态响应之间的矛盾,证明了基本自适应系统是一个关于中心角频率对称的陷波器.理论推导证明了由于系统权向量不为一个恒定的值,导致该系统不能准确提取谐波电流.在此基础上提出了一种改进型模拟自适应对消检测方法,讨论了改进型自适应系统的稳...     

Novel combination schemes of individual weighting factors sign subband adaptive filter algorithm

To improve the performance of the recently presented individual weighting factors sign subband adaptive filter (IWF‐SSAF) algorithm, its 2 combination algorithms using different step sizes are proposed. The first algorithm is to convexly combine the weight vectors of a large step‐size IWF‐SSAF filter and a small step‐size one; and the second algorithm is to obtain a time‐varying step size for the IWF‐SSAF by combining a large step size and a small one. The minimization of the sum of the l₁‐norm of subband errors is used to indirectly update the mixing parameters in these 2 algorithms through a modified sigmoidal function. Moreover, in the first algorithm, to implement a smooth transition from the large step‐size IWF‐SSAF filter to the small step‐size one, the component filters receive a cyclic feedback of the combined weight vector. Both proposed algorithms have almost the same convergence performance, but the second algorithm saves computational cost. Simulation results in impulsive noise scenarios demonstrate the superiority of our proposed algorithms.     

Two-Pole and Multi-Pole Notch Filters: A Computationally Effective Solution for GNSS Interference Detection and Mitigation

In a global navigation satellite system (GNSS) receiver, the presence of detection and mitigation units, capable of reducing the impact of disturbing signals, can extremely enhance the position accuracy. However, the presence of such units is usually limited to professional receivers that dispose of additional computational power that can be used for interference detection and mitigation. In this paper, the two-pole notch filter, that is the natural extension of the one-pole notch filter, is proposed as computationally effective solution for interference detection and mitigation. The notch filter structure and the adaptive algorithm employed for tracking the disturbing signal are analyzed, and an interference detection unit, based on the adaptive algorithm convergence, is proposed. The two-pole notch filter coupled with the detection unit is used as elementary block for the design of a multi-pole notch filter that can efficiently mitigate more than one CW interference. Theoretical and simulative analyses show the feasibility and the good performance of the proposed method.     

基于自适应陷波器的电网频率估计方法

本文针对电网频率难以精确测量的问题,提出利用自适应陷波器结构的滤波功能,结合电网电压基波信号及其谐波信号的特性,以陷波器实时输出平方值作为陷波器参数的优化目标函数,根据随机优化理论推导出陷波器参数在线迭代算法,从而对与频率有关的陷波参数进行自适应调整,达到电网频率估计的目的。该方法结构简单,通过仿真分析和实验测试,检验了所提出的频率估计算法的收敛速度和逼近精度,表明基于自适应陷波器的电网频率估计方法,可以准确估计出电网频率。     

一种基于自适应陷波器的电网频率测量新方法

该文提出一种测量和跟踪电网频率变化的新方法：采用两级自适应陷波滤波器结构，第一级陷波器用来滤除谐波并得到加强的基波成分，再经过降采样处理把基波频谱拓宽后，由第二级陷波器来估计基波频率。陷波器的自适应算法是非梯度搜索的，使其运算量大大简化，适用于实时处理的场合。并通过仿真来说明该方法具有测量精度高和响应时间快的特点。     

基于SOGI-FLL的伺服传动系统谐振频率检测方法

在工业伺服控制领域,常采用陷波滤波器抑制机械谐振,而准确地获取谐振频率是谐振成功抑制的首要条件。采用二阶广义积分器-锁频环(SOGI-FLL)分析速度控制器输出信号以检测谐振频率。首先介绍了SOGI-FLL的基本结构及锁频环的工作原理,然后分析了幅值频率自适应SOGI-FLL的频率响应性能,最后通过仿真和试验验证了幅值频率自适应SOGI-FLL能准确且快速地测出谐振频率,将所测频率用于陷波滤波器参数设置,成功抑制了机械谐振。     

基于自适应陷波器的科氏流量计信号频率跟踪新方法

为提高对科氏流量计信号频率随机缓慢变化的持续跟踪能力,以对格型ANF、简化格型ANF和基于SMM的新式ANF三种典型自适应陷波器的性能比较分析为基础,提出了一种科氏流量计信号频率跟踪新方法。该方法对频率、幅值和相位均随机游动变化的科氏流量计信号,首先采用新式ANF快速检测信号频率并作短时频率跟踪,待其收敛后简化格型ANF开始并行工作,在简化格型ANF收敛后取代新式ANF持续跟踪信号频率的变化。仿真结果表明,本文方法比格型ANF方法收敛速度更快、频率跟踪精度更高,比单一采用新式ANF方法计算更为简便,是一种科氏流量计信号处理的有效方法。     

基于系数查找表的全相位FIR自适应陷波器

本文分析了全相位DFT数字滤波器优良的频域传输特性,进而提出基于滤波器系数抽取查找表的全相位自适应陷波算法.该算法以陷波器输出信号的能量作为判断依据选择抽头系数,省去了传统频域FIR滤波器的FFT步骤,因而计算量小,而且变传统闭环控制为开环控制,因而收敛快,并可把频率设置点上的干扰完全陷波到0.因而相比于传统ⅡR和FIR陷波器,性能有了很大提高.     

Filter proportionate normalized least mean square algorithm for a sparse system

In this paper, the proportionate normalized least mean square (PNLMS) and its modifications, such as improved PNLMS (IPNLMS) and μ‐law PNLMS (MPNLMS) algorithms, developed for a sparse system, are analyzed for a compressed input signal. This analysis is based on a comparative study of the steady‐state error and convergence time for the original signal and the compressed signal. Further, in this paper, a filter PNLMS (FPNLMS) algorithm that is a modification of the IPNLMS algorithm is proposed. The FPNLMS algorithm uses a step size varying in time to adapt to the sparse system. Simulations are carried out to compare the proposed FPNLMS algorithm for different signal‐to‐noise ratio for a compressed input signal with existing algorithms, ie, PNLMS, MPNLMS, and IPNLMS algorithms. The FPNLMS algorithm achieves a better steady‐state and convergence time compared with other existing algorithms in both low and high SNRs. The FPNLMS algorithm is further simulated for a real transfer function to show its robustness compared with existing algorithms.     

基于DDPG的柔性伺服系统级联陷波器设计

针对柔性伺服系统的多频谐振抑制问题,提出一种基于DDPG的级联陷波器参数整定方法。以系统速度环开环bode图及陷波器bode图预处理结果作为训练数据,并以相位裕度作为奖励函数训练神经网络,实现所设计的伺服系统级联陷波器深度及宽度参数优化训练。搭建了三质量柔性伺服系统实验平台,并开展了多频谐振抑制实验,实验结果表明所提出的参数选择方法能够 找到具有最大相位裕度的陷波器参数,并有效地抑制系统多频谐振。     

Multiband sound source localization algorithm for directional enhancement in hearing aids

In order to improve speech comprehension for hearing‐impaired patients under complex scenarios, directional enhancement algorithm based on the sound location will be one of the future research directions. A multiband sound localization algorithm based on auditory bionics is proposed in this paper. First, according to the property of cochlear frequency‐division and hearing‐masking characteristic, the sound is divided into multiband signals. Then, the interaural time differences (ITDs) of the sensitive frequency bands are taken as the localization cues. To improve the location precision, the real ITD is extracted based on the Haas effect. Finally, the HRTF (head‐related transfer function) model is used to convert the ITD to the sound position. To improve the robustness of the algorithm, a multiband weighted strategy is adopted. In addition, a parameter adjustment method is proposed to effectively fit the hearing aid. Simulation and scenario experiments show that the algorithm has strong anti‐interference and high location accuracy. Furthermore, it is designed based on the existing algorithms for hearing aids and has low computation, so it is very suitable for hearing aids and other low‐power real‐time devices. © 2016 Institute of Electrical Engineers of Japan. Published by John Wiley & Sons, Inc.     

Analytical synthesis of voltage‐mode even/odd‐nth‐order differential difference current conveyor and fully differential current conveyor II‐grounded resistor and capacitor universal filter structures

A complete definition of an odd/even‐nth‐order notch or band‐reject filter transfer function is presented. Based on the differences between the input voltage and (i) an nth‐order high‐pass; (ii) a traditional nth‐order notch; and (iii) an nth‐order all‐pass filtering transfer function, a systematic method has been proposed to derive a universal filter structure that can realize voltage‐mode odd/even‐nth‐order low‐pass, band‐pass, high‐pass, all‐pass and traditional notch filters. The intrinsic capability of voltage‐mode addition and subtraction of the two active elements, differential difference current conveyors and fully differential current conveyors, is used to advantage in the aforementioned synthesis procedure. Based upon the definition of an nth‐order notch or band‐reject filter transfer function proposed in this paper, the aforementioned universal one has been further extended to the newly defined nth‐order band rejection filter. The voltage and current tracking errors of the two active elements are compensated by varying the resistances of the proposed filter. Filtering feasibility, stability, component sensitivities, linear and dynamic ranges, power consumption, and noise are simulated using H‐Spice with 0.35 µm process. Compared to some of the recently reported universal biquads, the new one is shown to enjoy the lowest component sensitivities and the best output accuracy for all‐pass signals. Moreover, Monte Carlo and two‐tone tests for intermodulation linearity simulations are also investigated. Copyright © 2014 John Wiley & Sons, Ltd.     

摆臂高频往复旋转定位抑振控制

为实现高加速度高频往复旋转摆臂的高精快速定位控制,提出多种振动抑制方案。包括使用S型加减速控制以减小由摆臂惯性力造成的振动,降低摆臂转动惯量和提高摆臂联结以防止机械谐振发生,以及伺服滤波技术。在伺服滤波技术中,使用陷波滤波器抑制机械谐振,优化低通滤波器保证系统稳定性,以及使用一种新的带阻滤波器抑制中低频振动。最终实现了旋转摆臂的振动控制并达到了高精定位要求并成功应用于LED分选设备中。