用于拖曳式线列阵的一种新的线谱增强系统

拖曳式线列阵是近来声呐信号处理领域中非常活跃的课题.自适应线谱增强器有可能用于拖曳式线列阵的信号处理系统.在利用LMS迭代算法的自适应线谱增强器中,由于存在迭代噪声,系统增益与输入信噪比有关.如果不抑制拖曳平台的噪声,线谱增强系统的效用就无法发挥出来,文中提出用自适应噪声抵消系统和自适应线谱增强器级联的概念.首先把拖曳平台的噪声作为干扰加以抑制,再对输出信号进行线谱增强,从而可以提取微弱信号的线谱分量.     

存在声反馈的前馈有源噪声控制系统性能分析

在一些应用场合,前馈有源噪声控制系统中次级源产生的声信号会反馈至参考传声器,影响参考信号质量和系统稳定,导致控制性能下降。引入了等效次级路径的概念,并通过等效次级路径与实际路径的相位偏差分析存在声反馈时的收敛性能。若某些频率的相位偏差大于90°,则这些频率附近将较难收敛,降噪性能下降,甚至导致系统不稳定。通过仿真和实验对单指向传声器声学方法、自适应滤波u型最小均方差(FuLMS)算法、反馈中和算法和在线建模算法共4种解决声反馈问题的方法的性能进行了比较。结果表明,4种方法都能提高存在声反馈时的前馈有源噪声控制系统的性能,有效解决声反馈引起的问题,但各有优缺点。单指向传声器方法最为方便,但低频指向性较差。FuLMS算法运算量较低,但不能保证收敛。反馈中和算法性能最好,但当系统时变时鲁棒性较差。在线建模算法不需要额外滤波器,但由于参数调节复杂,降噪性能稍差。      

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。     

听觉频域掩蔽效应的自适应β阶贝叶斯感知估计语音增强算法

为了在噪声抑制和语音失真中之间寻找最佳平衡,提出了一种听觉频域掩蔽效应的自适应β阶贝叶斯感知估计语音增强算法,以期提高语音增强的综合性能。算法利用了人耳的听觉掩蔽效应,根据计算得到的频域掩蔽阈自适应调整β阶贝叶斯感知估计语音增强算法中的β值,从而仅将噪声抑制在掩蔽阈之下,保留较多的语音信息,降低语音失真。并分别用客观和主观评价方式,对所提出的算法的性能进行了评估,并与原来基于信噪比的自适应β阶贝叶斯感知估计语音增强算法进行了比较。结果表明,频域掩蔽的β阶贝叶斯感知估计方法的综合客观评价结果在信噪比为-10 dB至5 dB之间时均高于基于信噪比的自适应β阶贝叶斯感知估计语音增强算法。主观评价结果也表明频域掩蔽的β阶贝叶斯感知估计方法能在尽量保留语音信息的同时,较好的抑制背景噪声。      

用于块稀疏系统辨识的混合范数约束LMS算法

<正>系统辨识是自适应滤波算法的一个重要应用场景。传统自适应滤波算法只是利用已知的输入信号和输出信号,通过迭代更新的方式估计未知系统向量。然而,在实际的应用场景中,未知系统自身的结构特性(例如块稀疏性)可以被用于改善算法的收敛性能。系统具有块稀疏特性是指,未知系统的时域脉冲响应包含了大量零值或者接近零值的元素,其余为显著值,且其分布较为集中。在会议系统中,电话线路的回声路径就是典型的单簇结构稀疏系统;卫星通信线路     

主瓣约束自适应旁瓣抵消器的实验研究

空间预滤波加无约束自适应噪声抵消器构成了一种主瓣约束的自适应旁瓣抵消器。这种自适应旁瓣抵消器由于实现了保持主瓣方向以及主瓣形状的导数约束，具有较好的主瓣保持性能，且结构简单，便于工程实现。本文在原有理论研究的基础上，着重从实验方面研究这种自适应旁瓣抵消器的主瓣保持和旁瓣抵消性能。设计并构造了一个实验系统，该系统利用一种新的空间预滤波结构，使其更宜于工程实现。进行了水池和海上试验，详细考察了输入信噪比、信号干扰比、信号与干扰带宽和自适应步长因子等参数对系统性能的影响。取得的结果证实了理论分析的正确性。     

基于近场聚焦逆波束形成的UUV平台噪声自适应抵消

UUV平台自噪声会对舷侧阵产生干扰，从而影响其探测性能，本文针对平台尾部自噪声源的直达噪声和海面反射噪声，提出了基于近场聚焦逆波束形成的平台噪声自适应抵消方法。文中利用UUV平台设计参数、深度以及运动姿态信息对尾部自噪声传播到舷侧阵的路径建立几何模型，并分析了影响该路径的因素；通过该模型，平台尾部自噪声能够实时自适应地通过聚焦波束形成被估计出来，从而在舷侧阵的接收信号中实现干扰抵消。仿真结果与海试实验数据处理结果表明，该方法能够有效抑制尾部自噪声干扰，显著提高UUV探测弱目标的能力。     

基于水声信道传播时延排序的分层空时信号检测

基于分层空时编码的多输入多输出技术是一种极具潜力的高速水声通信技术, 但要实现这种潜力需要复杂的空时信号处理方法, 以抵消来自水声信道的多径干扰和异步到达干扰, 以及叠加在接收端的各层信号之间的干扰. 对低复杂度的空时信号处理方案进行了研究, 提出了一种基于子信道传播时延排序的有序连续干扰抵消信号检测算法, 利用子信道间的传播时延差, 实现可使差错概率最小的最佳检测排序; 给出了利用信道估计, 以极低的计算量确定排序的方法, 从而可以大幅降低信号检测的计算复杂度. 采用低复杂度的单载波频域均衡来抵消水声信道中的码间干扰和异步到达干扰. 仿真结果表明, 基于时延排序的信号处理算法可以获得检测性能的改善, 而且性能增益在高数据率时更加显著. 研究结果表明, 采用有效的信号处理方法可使水声信道中造成信号检测干扰的传播时延成为改善系统性能的有利因素.     

一种采用脉冲串的自适应有源干扰抵消器

在许多应用中，周期性干扰是由旋转的机械设备所产生的。一个周期性信号可以与一个特殊形式的周期性的脉冲串完全相干。本文提出了一种采用脉冲串的自适应有源噪声抵消器(AANC)来抵消周期性干扰，这种抵消器产生一个与干扰信号相同步的周期性的单位脉冲串作为参考输入。将这种有源噪声抵消器和Delayed-X LMS算法相结合，可以大大地减小运算量，并且也可以保证良好的收敛特性。理论分析和计算机仿真验证了本文的结果。     

一种维纳加权频域自适应时延估计算法

提出了一种基于频域自适应滤波的维纳加权时延估计算法。该算法首先实现了对输入信号的维纳自适应滤波预处理,然后在频域实现了自适应时延估计。该算法在将相位数据拟合时,做了一次加权处理,有效地改善了时延估计的精度和自适应收敛速度。最后用实测的直升机噪声数据进行了仿真,验证了新算法的有效性。     

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.     

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.     

Stereophonic channel decorrelation using a binaural masking model

Traditional methods often only use monaural masking models to decorrelate input signals for stereo acoustic echo cancellation. Whereas, it seems more reasonable to use binaural masking models for the following two reasons. First, stereo signals are heard by two ears rather than just one. Second, psychoacoustic researchers have already shown that there are obvious masking level differences between binaural masking models and monaural masking models. By studying binaural masking level difference models, we first introduce a simplified binaural masking model for stereo acoustic echo cancellation. Considering that the interaural time difference is dominant at low frequencies (?$?$1.5  kHz) and the interaural level difference is a major cue at higher frequencies, we propose to use different signal decorrelation schemes at these two frequency bands. In the low-frequency band, a pitch-driven sinusoidal injection scheme is proposed to maintain the interaural time difference, where the amount of injection is determined by the proposed binaural masking model. In the high-frequency band, a modified sinusoidal phase modulation scheme is applied to make a trade-off between preserving the interaural level difference and decorrelating the stereophonic input signals. Assessment results show that the proposed method can effectively improve the non-unique problem and retain good speech quality.     

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.     

ADAPTIVE NONLINEAR FEEDBACK CONTROL OF CHAOTIC SYSTEMS BASED ON REDUCED PARAMETER QUADRATIC PREDICTORS

An adaptive nonlinear feedback-control method is proposed to control continuous-time chaotic dynamical systems, where the adaptive nonlinear controller acts on only one-dimensional error signal between the desired state and the observed chaotic state of a system. The reduced parameter adaptive quadratic predictor used in adaptive feedback cancellation of the nonlinear terms can control the system to any desired state. Computer simulation results on the Lorenz system are shown to demonstrate the effectiveness of this feedback-control method.     

Generalized projective synchronization of chaotic systems with unknown dead-zone input: observer-based approach

In this paper we investigate the synchronization problem of drive-response chaotic systems with a scalar coupling signal. By using the scalar transmitted signal from the drive chaotic system, an observer-based response chaotic system with dead-zone nonlinear input is designed. An output feedback control technique is derived to achieve generalized projective synchronization between the drive system and the response system. Furthermore, an adaptive control law is established that guarantees generalized projective synchronization without the knowledge of system nonlinearity, and/or system parameters as well as that of parameters in dead-zone input nonlinearity. Two illustrative examples are given to demonstrate the effectiveness of the proposed synchronization scheme.     

霍尔传感器温度漂移补偿电路设计

霍尔传感器作为一种半导体器件,其灵敏度随着温度变化而产生漂移的特性,限制了其在高精度磁场测量场合的应用。传统的温度补偿方法虽然对温度变化的一次项进行了完全补偿,但是却引入了温度变化二次项的误差。因此,对于温度变化显著的高精度测量场合,传统温度补偿法将不再适用。设计了一种闭环反馈电路,通过温度传感器采集温度信号,与信号处理电路的最终输出信号进行运算后送回信号处理电路的输入端进行补偿,而并不是简单地将温度信号与霍尔信号的输入信号进行相加后送入信号处理电路。仿真分析结果表明,通过调节补偿电路的反馈比例系数与霍尔芯片温度漂移系数,可以完全补偿霍尔芯片的灵敏度漂移。因此,这种闭环补偿方法可以不引入与温度变化二次项有关的误差,消除因温度变化产生的漂移,不仅适用于霍尔传感器,也适用于其他会随着温度漂移的传感器。     

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.     

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。     

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.