应用于声反馈抑制的变步长归一化子带自适应滤波算法*

针对自适应滤波算法中稳态失调量和收敛速度之间的矛盾,提出了一种新的变步长归一化子带自适应滤波算法。该算法在系统噪声抵消原理的基础上,用迭代收缩的方法估计得到无噪先验子带误差的功率,对每个子带步长进行更新。对所提出的算法进行数学分析,可以得出该算法是稳定的和收敛的。在长回声路径和短回声路径两种情况下,将该算法应用于助听器声反馈抑制系统中。相对于其他归一化子带自适应滤波算法,仿真实验表明,所提算法实现了更快的收敛速度,获得了更低的失调量。     

基于干扰对消的红外焦平面非均匀性校正算法

红外焦平面器件的非均匀性产生机理复杂,难以准确拟合探测元响应曲线。提出了一种基于相关干扰抵消的非均匀性校正算法,以预先采集到的一帧黑体面源图像做为自适应干扰对消器的参考输入图像,自适应滤波器由参考输入图像迭代计算出待校正红外图像的空间噪声的最佳估计,实现从空间噪声中提取真实图像信号。自适应滤波算法采用变步长最小均方误差算法,减少了算法的运算量,提高了算法的收敛速度。理论分析以及针对实际红外图像的仿真结果表明,提出的算法校正效果好,收敛速度快,更易于工程实现。     

VHF波段变容管调谐式超导陷波器

设计制作了一款VHF波段变容管调谐式超导陷波器,该陷波器采用一种结构紧凑的电容加载的矩形回路谐振器,谐振器与主线之间的耦合形式为混合耦合。通过调节加载在谐振器上变容二极管的反偏电压,实现带阻滤波器中心频率可调。测试表明,陷波器中心频率可调范围为251~360MHz,陷波器中心频点抑制40dB,3dB带宽9dB,插入损耗0.5dB。     

应用于助听器反馈抑制的信噪比自适应滤波算法

针对基于自适应滤波器的助听器反馈抑制系统,该文提出了一种基于信噪比的归一化最小均方误差算法,采用最小值统计法估计误差信号的噪声分量,从而计算出误差信号的信噪比来计算自适应滤波系数的更新步长。当误差信号信噪比越高,语声占主要成分,信号的相关性越强,此时将滤波器的更新步长控制在较小值,减小滤波器的失调量;当信噪比越低时,噪声占主要成分,信号的相关性相对较弱,更新步长取较大值,加快滤波器的收敛速度。在仿真实验中,该文提出的基于信噪比的归一化最小均方误差算法相较于传统算法在平均稳态失调量和稳态失调范围上分别低1 dB和2 dB,其最大稳态增益提高了4 dB,同时具有更快的稳态收敛速度,验证了该文提出算法的有效性。     

变步长似p范数约束稀疏水声信道估计方法

针对范数约束类归一化最小均方(NLMS)算法在正交频分复用(OFDM)稀疏水声信道估计中误码率较高的问题,提出一种改进的变步长似p范数约束信道估计方法。采用改进双Logistic函数构造步长,并将误差信号自相关函数引入其中,实时调整步长和零吸引项,使得收敛速度和估计精度能够很好地折中。算法仿真结果表明,在浅海多径稀疏水声信道下,相比于传统方法,所提出的信道估计获得的最高性能提升为收敛速度提高72.3%,稳态误差降低95.9%。湖上试验数据处理结果显示,相比于传统信道估计方法,所提出的方法能使通信误码率降低2~3个数量级,实现零误码水声通信。      

双通道回声抵消系统中改进算法的定点化实现

本文研究了双通道回声抵消系统的三个核心模块:频域多延时自适应滤波器算法(MDF),双端说话检测算法(DTD),以及残留回声抑制算法(RES)。针对频域算法的特点提出了改进的双端说话检测算法和基于维纳滤波与谱减法的残留回声抑制算法,然后以上述三种算法为核心模块实现双通道回声抵消系统,并对系统定点化以便在定点DSP处理器上实时实现,分析并解决了定点化的精度问题所带来的影响。     

一种提高RS422通信BIT可靠性的设计方法

电动静液作动器是飞机操纵系统的关键部件,要求有较好的速度平稳性。系统内存在泄漏非线性和摩擦非线性等影响速度平稳性的因素。滑模控制可以有效抑制系统内非线性因素的影响,但是由于抖振现象的存在限制了速度平稳性的进一步提升。针对固定切换增益的滑模控制方法的不足,提出一种基于变结构滤波器的自适应滑模控制方法。采用变结构滤波器估计系统状态信息,估计的系统状态信息用于构建滑模面,采用自适应切换增益来导出控制率,有效减小了抖振幅度。仿真结果证明了自适应滑模控制方法的有效性,采用这种方法提高了电动静液作动器的速度平稳性。     

自适应线谱干扰抵消器的性能及其非维纳滤波现象

自适应滤波器的性能通常采用维纳滤波器理论来分析,本文指出自适应线谱干扰抵消器不符合经典解的现象。在对权系数的动态分析中,揭示了非维纳滤波效应产生的机理。     

多波束自适应噪声抵消法引论

本文给出按Wiener最小均方差准则设计的两种噪声抵消系统模型。给出计算输出信噪比的一般表达式,提出了把这种模型用于自适应波束成形时的实际检验准则。提出一种多波束自适应噪声抵消系统,分析了它的主要性能。本文的全部讨论都在频域上进行,文中给出利用信号平均功率谱密度和最佳线性滤波器的传输函数计算系统指向性的方法,给出了多波束噪声抵消法在自适应前后的指向性公式。实际的例子说明这种系统在抑制为数不多的干扰时具有很大的优越性。     

具有陷波电路结构的P波段低温低噪声放大器

该文的工作是设计和制作了一种具有陷波电路结构的P波段低温低噪声放大器。在低温75K环境下,工作频段为250-350MHz的范围内,该低温低噪声放大器具有优异的性能,噪声系数小于0.4dB,增益为14.4dB,增益平坦度小于0.05dB,输入反射损耗S11<-20dB,输出反射损耗S22<-20dB。同时在工作频段外的高温超导滤波器寄生通带内,该低温低噪声放大器成功实现了传输陷波响应,加强了系统对前端高温超导滤波器产生的寄生通带的衰减和抑制。     

A robust adaptive hybrid feedback cancellation scheme for hearing aids in the presence of outliers

It has been demonstrated that the Filtered-x Wilcoxon LMS (FxWLMS) based adaptive filter mitigates the effect of the outliers acquired by the microphone signal of hearing aids by minimizing the Wilcoxon norm and hence shows better cancellation performance than the existing Filtered-x LMS (FxLMS) algorithm. The prediction error method based adaptive feedback canceller (PEMAFC) reduces the bias present in the estimate of the feedback path due to the continuous adaptive filtering (CAF). However, the impulse response of the measured feedback path is close to zero for the first many samples due to the delay introduced by ADC converters and then contains few significant values, which results in slow convergence rate when an adaptive filter is used to model the same. To overcome this limitation, we propose a proportionate normalized WLMS (PNWLMS) algorithm based PEMAFC (P-PNWLMS) for feedback cancellation in hearing aid in the presence of outliers. Further, with an objective to improve the convergence rate and performance accuracy simultaneously, this paper proposes a novel convex PNWLMS (CPNWLMS) algorithm which incorporates convex combination of PNWLMS and WLMS algorithms. The weight update equations are derived for PEMAFC trained by PNWLMS (P-PNWLMS) and CPNWLMS (P-CPNWLMS) algorithms respectively. The results of the simulation study show improved performance of the proposed CPNWLMS based adaptive filter over its component filters.     

Constrained adaptation for feedback cancellation in hearing aids.

In feedback cancellation in hearing aids, an adaptive filter is used to model the feedback path. The output of the adaptive filter is subtracted from the microphone signal to cancel the acoustic and mechanical feedback picked up by the microphone, thus allowing more gain in the hearing aid. In general, the feedback-cancellation filter adapts on the hearing-aid input signal, and signal cancellation and coloration artifacts can occur for a narrow-band input. In this paper, two procedures for LMS adaptation with a constraint on the magnitude of the adaptive weight vector are derived. The constraints greatly reduce the probability that the adaptive filter will cancel a narrow-band input. Simulation results are used to demonstrate the efficacy of the constrained adaptation.     

Room reverberation effects in hearing aid feedback cancellation

Room reverberation can affect feedback cancellation in hearing aids, with the strength of the effects depending on the acoustical conditions. These effects were studied using a behind the ear (BTE) hearing aid mounted on a dummy head and coupled to the ear canal via an open fitting. The hearing aid impulse response was measured for the dummy head placed at eight closely spaced locations in a typical office. The feedback cancellation in the hearing aid used a set of filter coefficients that were initialized for one location within the room, and then allowed to adapt to the feedback path measured at the same or to a different location. The maximum stable gain for the hearing aid was then estimated without feedback cancellation, for the initial set of feedback cancellation filter coefficients prior to adaptation, and for the feedback cancellation filter after adaptation. A low-order ARMA model combining a fixed set of poles with an adaptive FIR filter is shown to be effective in representing the feedback path exclusive of reverberation. Increasing the adaptive filter length has only a small benefit in improving the feedback cancellation performance due to the inability of the system to model the room reverberation. The mismatch between the modeled and actual feedback paths limits the headroom increase that can be achieved when using feedback cancellation, and varies with the location within the room.     

一种基于粒子滤波的雷达抗欺骗干扰方法

针对雷达的抗欺骗干扰问题,利用欺骗干扰信号强于目标信号的特点,基于粒子滤波进行了抗欺骗干扰研究。当存在欺骗干扰时,粒子滤波中各粒子的重要性权值会明显减小,据此可以检测干扰,并对受到干扰的数据点进行置零处理,使得欺骗干扰不再与匹配滤波器匹配,从而达到抑制干扰的目的。设计的粒子滤波算法不需要估计系统状态转移函数、系统量测噪声,从而使得算法更具实用性。仿真结果表明,该算法能有效地抑制欺骗干扰,且对干信比不敏感。     

Theoretical and experimental study of Chen chaotic system with notch filter feedback control

Since the past two decades, the time delay feedback control method has attracted more and more attention in chaos control studies because of its simplicity and efficiency compared with other chaos control schemes. Recently, it has been proposed to suppress low-dimensional chaos with the notch filter feedback control method, which can be implemented in a laser system. In this work, we have analytically determined the controllable conditions for notch filter feedback controlling of Chen chaotic system in terms of the Hopf bifurcation theory. The conditions for notch filter feedback controlled Chen chaoitc system having a stable limit cycle solution are given. Meanwhile, we also analysed the Hopf bifurcation direction, which is very important for parameter settings in notch filter feedback control applications. Finally, we apply the notch filter feedback control methods to the electronic circuit experiments and numerical simulations based on the theoretical analysis. The controlling results of notch filter feedback control method well prove the feasibility and reliability of the theoretical analysis.     

Adaptive feedback cancellation in hearing aids with clipping in the feedback path

Adaptive linear filtering algorithms are commonly used to cancel feedback in hearing aids. The use of these algorithms is based on the assumption that the feedback path is linear, so nonlinearities in the feedback path may affect performance. This study investigated the effect on feedback canceller performance of clipping of the feedback signal arriving at the microphone, as well as the benefit of applying identical clipping to the cancellation signal so that the cancellation path modeled the nonlinearity of the feedback path. Feedback signal clipping limited the amount of added stable gain that the feedback canceller could provide, and caused misadjustment in response to high-level inputs, by biasing adaptive filter coefficients toward lower magnitudes. Cancellation signal clipping mitigated these negative effects, permitting higher amounts of added stable gain and less misadjustment in response to high-level inputs, but the benefit was reduced in the presence of the highest-level inputs.     

Design and analysis of a BLPC vocoder-based adaptive feedback cancellation with probe noise

The band-limited linear predictive coding (BLPC) vocoder-based adaptive feedback cancellation (AFC) removes the high-frequency bias, while the low frequency bias persists between the desired input signal and the loudspeaker signal in the estimate of the feedback path. In this paper, we present a BLPC vocoder-based adaptive feedback canceller with probe noise with an objective of reducing the low-frequency bias in digital hearing-aids. A step-wise mathematical analysis of the proposed feedback canceller is presented employing the recursive least square and normalized least mean square adaptive algorithms. It is observed that the optimal solution of the feedback path is unbiased for an unshaped probe noise, but is biased for a shaped probe signal; the bias term does not consist of correlation between the desired input and the loudspeaker output. The identifiability conditions are analysed and it is shown that a delay, greater than or equal to the length of the adaptive filter, must be introduced in the forward path to achieve an unbiased feedback path estimate. Algorithm analysis and computer simulations presented in this paper justify the reason for selecting the proposed design over the existing BLPC vocoder-based feedback cancellation algorithm.     

Robust Blind Adaptive Channel Equalization in Chaotic Communication Systems

Based on the bounded property and statistics of chaotic signal and the idea of set-membership identification, we propose a set-membership generalized least mean square （SM-GLMS） algorithm with variable step size for blind adaptive channel equalization in chaotic communication systems. The steady state performance of the proposed SM-GLMS algorithm is analysed, and comparison with an extended Kalman filter （EKF）-based adaptive algorithm and variable gain least mean square （VG-LMS） algorithm is performed for blind adaptive channel equalization. Simulations show that the proposed SM-GLMS algorithm can provide more significant steady state performance improvement than the EKF-based adaptive algorithm and VG-LMS algorithm.     

数字信号处理器红外弱小目标搜索算法

提出了一种基于数字信号处理器（DSP）的红外弱小目标搜索高速实时算法。算法主要针对天空中不规则云团在预处理过程中残留的边缘而设计,是在背景预测算法基础上改进的一种基于陷波滤波器的搜索算法。在DSP实时处理平台上实现并经实际场景试验验证,该算法能有效削弱残留边缘的影响,使最远探测距离指标上升至前向迎头26 km,实时处理速度达到75 Hz,满足系统要求。采用信噪比增益指标对算法进行了评价,结果显示,该算法能够显著提高图像信噪比,利于检测出目标。