小波尺度相关滤波及其改进算法在电火花声源信号处理中的应用

分析提取宽带电火花声源信号,受到船舶辐射噪声的严重干扰,在5 kHz以下的能量集中区信噪比低,且二者Lipschitz指数特性相近,传统基于相邻尺度相关的滤波算法抗干扰能力有限。根据宽带电火花声源信号不同频带所受干扰的不同,信噪比较高的小尺度高频小波系数,采用相邻尺度相关的滤波算法;信噪比较低的大尺度中低频小波系数,采用跨尺度相关的滤波算法,并对算法中阈值系数的选取方法进行修正。结果表明,该算法滤波效果良好,有效的提取了电火花声源信号,适合窄带强干扰背景噪声下的宽带水声信号处理。      

一种数字有源降噪耳机控制器设计方法

为了得到一种作用于低频宽带噪声的有源降噪耳机,采用固定系数的数字控制器实施降噪,并提出了一种前反馈混合的控制器设计方法。比较了前馈及混合两种控制模式下对低频宽带噪声的控制效果;进一步提出了一种综合优化方法,改善噪声不同角度入射下的降噪性能。仿真和基于DSP平台的实验表明,提出的方法对多方向入射的20～1500 Hz低频宽带噪声均具有较好控制效果。     

小波包分解域的超声图像非同态滤波降噪算法

针对医学超声图像中固有的斑点噪声,提出了一种基于小波包变换和非同态滤波的超声图像降噪算法,通过对小波包子带系数的邻域相关性进行研究,求解子带系数的归一化自协方差函数进而得到独立于小波包系数的邻域算子,提高了算法对信号和噪声的分离能力.然后对邻域算子和小波包系数进行最小均方误差联合估计得到非同态滤波降噪方程,实现对超声图...     

受污染混沌信号的协同滤波降噪

根据混沌吸引子的自相似分形特性,提出了一种利用协同滤波重构受污染混沌信号的降噪算法.所设计的降噪算法通过对相似片段的分组将一维混沌信号的降噪转化为一个二维联合滤波问题;然后,在二维变换域用阈值法衰减噪声;最后,通过反变换获得原始信号的估计.由于分组中的相似片段具有良好的相关性,与直接在一维变换域做阈值降噪相比,分组的二维变换能获得原信号更稀疏的表示,更好地抑制噪声.仿真结果表明,该算法对原始混沌信号的重构精度和信噪比的提升都优于小波阈值、局部曲线拟合等现有的混沌信号降噪方法,对相图的还原质量也更好.     

基于双传声器的蓝牙耳机降噪算法

用于免提通信设备的语音增强算法一直是研究的热点问题,而算法处理结果的音质问题近年来也备受关注。针对基于双传声器降噪的蓝牙耳机系统,将常用多通道传声器降噪算法归纳为基于相干函数法和基于空间预分离法这两大类进行分析和比较。基于相干函数法利用两个通道间信号的相干函数对含噪信号滤波达到降噪目的,而基于空间预分离法利用空间特性从含噪信号中分离出噪声参考信号来消除噪声。分析基于降噪量、语音音质和综合性能三个指标,从约束语音损伤的角度分析最优解的形式,并对比两类算法的实际性能。结果表明选择合适的算法可权衡降噪量与语音损伤,达到较好的综合性能。     

调节水床效应的双梯度有源噪声控制自适应算法*

为了使自适应反馈有源噪声控制系统能够兼顾水床效应,通过约束次级信号的能量以调节噪声放大,设计了一种双梯度算法。当次级信号满足约束,算法沿着最小化误差信号能量的梯度方向迭代,反之,则沿着最小化次级信号能量的方向迭代。在有源降噪耳机实例中的对比结果表明,该算法能调节噪声放大并保持较大的降噪带宽,且运算量没有显著增加。     

阵列信号源中的窄带噪声实时生成与时延控制技术研究

针对鱼雷声自导测试信号源中的目标辐射信号模拟问题,集中探讨窄带噪声的生成及其通道间时延控制算法。首先研究窄带噪声实时生成算法,基于线性同余法生成均匀噪声,以其作为查询噪声池的地址,产生宽带噪声,并通过带通滤波得到窄带噪声。为保证模拟目标的方位角精度,精确控制各个通道的时延量,提出适用于宽带信号的延时滤波算法,基于多采样率信号处理技术,通过对噪声信号进行插值、延迟、选抽、滤波等操作,提高时延控制精度。结合算法原理分析,得到延时滤波的简化算法。数字仿真表明,提出算法适用于窄带信号实时生成,时延控制精度高。     

改进粒子滤波在汽轮机故障诊断中的应用

针对汽轮机的振动信号容易受到较为复杂的随机噪声污染,提出了一种改进粒子滤波的振动信号降噪方法;首先建立采集振动信号的数学模型,将其作为粒子滤波的状态方程;然后利用小波分析提取采集振动信号的背景噪声,将其和状态信号一起作为观测信号,得到观测方程,把降噪问题转化成在状态空间模型下的滤波问题;由于采用序贯重要性采样的粒子滤波存在着样本退化问题,在重采样阶段采用了一种权值排序、优胜劣汰的重采样算法,就是对各粒子的归一化权值从小到大的排列顺序,并根据权值方差大小淘汰粒子,从而得到了改进的粒子滤波算法,在一定程度上解决了标准粒子滤波的退化问题;进而运用改进粒子滤波算法对振动信号进行降噪处理,降噪前信号和降噪后信号分别通过小波包分解系数求取频带能量,根据各个频带能量的变化提取故障特征向量浓缩了汽轮机振动故障的全部信息,对提取的故障特征向量应用诊断识别算法进行故障模式识别;通过对比降噪前信号和降噪后信号的故障诊断识别率,证明了改进粒子滤波在汽轮机故障诊断中的应用效果更佳。     

非高斯冲激干扰下基于Softplus函数的核自适应滤波算法

针对非高斯环境下一般自适应滤波算法性能严重下降问题,本文提出了一种基于Softplus函数的核分式低次幂自适应滤波算法(kernel fractional lower algorithm based on Softplus function,SP-KFLP),该算法将Softplus函数与核分式低次幂准则相结合,利用输出误差的非线性饱和特性通过随机梯度下降法更新权重.一方面利用Softplus函数的特点在保证了SP-KFLP算法具有良好的抗脉冲干扰性能的同时提高了其收敛速度;另一方面将低次幂误差的倒数作为权重向量更新公式的系数,利用误差突增使得权重向量不更新的方法来抵制冲激噪声,并对其均方收敛性进行了分析.在系统辨识环境下的仿真表明,该算法很好地兼顾了收敛速度和跟踪性能稳定误差的矛盾,在收敛速度和抗脉冲干扰鲁棒性方面优于核最小均方误差算法、核分式低次幂算法和S型核分式低次幂自适应滤波算法.     

改进粒子滤波在汽轮机故障诊断中的应用夏飞,郝硕涛

针对汽轮机的振动信号容易受到较为复杂的随机噪声污染,提出了一种改进粒子滤波的振动信号降噪方法。首先建立采集振动信号的数学模型,将其作为粒子滤波的状态方程;然后利用小波分析提取采集振动信号的背景噪声,将其和状态信号一起作为观测信号,得到观测方程,把降噪问题转化成在状态空间模型下的滤波问题。由于采用序贯重要性采样的粒子滤波存在着样本退化问题,在重采样阶段采用了一种权值排序、优胜劣汰的重采样算法,就是对各粒子的归一化权值从小到大的排列顺序,并根据权值方差大小淘汰粒子,从而得到了改进的粒子滤波算法,在一定程度上解决了标准粒子滤波的退化问题。进而运用改进粒子滤波算法对振动信号进行降噪处理,降噪前信号和降噪后信号分别通过小波包分解系数求取频带能量,根据各个频带能量的变化提取故障特征向量浓缩了汽轮机振动故障的全部信息,对提取的故障特征向量应用诊断识别算法进行故障模式识别。通过对比降噪前信号和降噪后信号的故障诊断识别率,证明了改进粒子滤波在汽轮机故障诊断中的应用效果更佳。     

一种基于虚拟传感的无需误差传声器的自适应有源降噪方法*

该文基于虚拟传感技术引入了一种用于耳机的无需误差传声器的自适应有源降噪方法。该算法仅使用一个参考传声器实现了一种前馈和反馈自适应算法结合的有源降噪算法,提高了有源降噪稳定性,简化了耳机硬件结构。利用DSP平台实现了该文提出的方案,并通过实验验证了其良好的降噪性能和实用价值。     

Causality study on a feedforward active noise control headset with different noise coming directions in free field

A systematic analysis is proposed to predict the performance of a typical feedforward single channel ANC headset in terms of the delay, especially the non-causal delay caused by different noise coming directions. First, the performance of a non-causal feedforward system for a band-limited noise is analyzed by using a simplified pure delay model, where it is found that the noise reduction bandwidth is narrowed and the maximum noise reduction is decreased with the increase of the non-causal delay. Second, a systematic method is developed, which can be used to predict the system performance with measured primary and secondary path transfer functions in most practical sound fields and to study the effects of the control filter length and the path delay on the performance. Then, the causality of a typical feedforward active noise control headset with the primary source at 0° and 90° positions in an anechoic chamber is analyzed, and the performance for the two locations predicted by the systematic analysis is shown in good agreements with the experiment results. Finally, an experiment of a typical feedforward active noise control headset in a reverberation chamber is carried out, which shows the validity of the proposed systematic analysis for other more practical sound fields.     

A robust filtered-s LMS algorithm for nonlinear active noise control

The performance of a nonlinear active noise control (ANC) system based on the recently developed filtered-s least mean square (FsLMS) algorithm deteriorates when strong disturbances in the ANC system are acquired by the microphones. To surmount this shortcoming, a novel robust FsLMS (RFsLMS) algorithm is proposed for a functional link artificial neural network (FLANN) based ANC system. The new ANC system is least sensitive to such disturbances and does not call for any prior information on the noise characteristics. The results obtained from simulation study establish the effectiveness of this new ANC scheme.     

多通道FxAP算法的电动汽车路噪控制

汽车噪声控制是主动噪声控制领域中经典的问题.伴随着电动化的普及,路噪控制将取代内燃机噪声控制,逐渐成为汽车噪声控制的主要领域.针对传统车内路噪控制系统适应性差、算法收敛速度慢、降噪量不足的问题,该文使用了多通道滤波-x仿射投影(FxAP)算法加快收敛速度,从而实现对噪声的快速追踪并控制.通过搭建了车内多通道头靠噪声控制...     

Active control of the volume acquisition noise in functional magnetic resonance imaging: method and psychoacoustical evaluation.

Functional magnetic resonance imaging (fMRI) provides a noninvasive tool for observing correlates of neural activity in the brain while a subject listens to sound. However, intense acoustic noise is generated in the process of capturing MR images. This noise stimulates the auditory nervous system, limiting the dynamic range available for displaying stimulus-driven activity. The noise is potentially damaging to hearing and is distracting for the subject. In an active noise control (ANC) system, a reference sample of a noise is processed to form a sound which adds destructively with the noise at the listener's ear. We describe an implementation of ANC in the electromagnetically hostile and physically compact MRI scanning environment. First, a prototype system was evaluated psychoacoustically in the laboratory, using the electrical drive to a noise-generating loudspeaker as the reference. This system produced 10-20 dB of subjective noise-reduction between 250 Hz and 1 kHz, and smaller amounts at higher frequencies. The system was modified to operate in a real MR scanner where the reference was obtained by recording the acoustic scanner noise. Objective reduction by 30-40 dB of the most intense component in scanner noises was realized between 500 Hz and 3500 Hz, and subjective reduction of 12 dB and 5 dB in tests at frequencies of 600 Hz and at 1.9 kHz, respectively. Although the benefit of ANC is limited by transmission paths to the cochlea other than air-conduction routes from the auditory meatus, ANC achieves worthwhile attenuation even in the frequency range of maximum bone conduction (1.5-2 kHz). ANC should, therefore, be generally useful during auditory fMRI.     

Binaural active noise control using parametric array loudspeakers

This paper reports the binaural active noise control (ANC) system developed to deal with factory noise. The control points are located in the vicinity of the left and right ears of a worker sitting along the production line. Due to the complicated safety requirements in the factory, secondary sources and error microphones are not allowed to be placed near the worker. Therefore, the proposed ANC system employs the feedforward structure and adopts the parametric array loudspeakers (PALs) as the secondary sources. The PAL is a type of directional loudspeaker that generates a much narrower sound field as compared to the conventional loudspeaker. Once the proposed ANC system has been trained offline, the error microphones can be removed. The performance of the binaural ANC system is successfully demonstrated based on a digital signal processor (DSP) implementation.     

Enhancement of noise reduction efficiency based on compensation in the ANC headset using fixed-point DSP

In this paper, we have proposed a new algorithm considering commutation error and feedback effect to enhance the convergence rate and noise reduction efficiency of ANC controller. In order to improve noise reduction performance of the ANC headset with fixed-point DSP, we have proposed a new FxLMS algorithm, FxLMS CF, which considers the commutation error and feedback. Also, using a non-real-time simulation, we have decided the phase and amplitude compensation factors of anti-noise signal considering round-off and quantization error, nonlinear distortion and delay of analog device. We estimated the phase and amplitude compensation factors by simulation without using any special measuring devices or analysis devices, and reduced the broadband noise by 24 dB.     

A hybrid active noise controller for finite ducts

A hybrid active noise controller (ANC) is proposed to solve some existing problems, which are related to the non-minimum phase (NMP) path models between uncollocated sensors and actuators in many ANC systems. For hybrid ANC schemes, the NMP path causes design difficulties to both feedforward and feedback control. These problems can be solved effectively by adding an extra actuator in the ANC system. A new design procedure is presented to take the greatest advantage of the extra actuator. Theoretical analysis and experimental results are presented to show the improved performance of the proposed ANC.     

Design and experimental tests of active control barriers for low-frequency stationary noise reduction in urban outdoor environment

Active noise control (ANC) techniques are based on the emission of an antiphase signal in order to cancel the noise produced by a primary source. ANC has been successfully applied especially for reducing noise in confined environments, such as headphones and ducts. In this study, we present an application of ANC concepts to the design of an anti-noise barrier for an outdoor environment and its experimental testing. Even though passive techniques are effective in noise reduction at middle-high frequencies, they become less efficient at low frequencies (below 300 Hz) due to the limited dimensions of commonly deployable barriers. In this paper, we analyze the properties of a low-cost active noise system able to efficiently operate on stationary, almost pure-tone, low-frequency noise, such as that produced by electrical transformers and reactors in power and transformation plants. A prototype has been implemented and on-the-field experimental tests have been carried out. The results (confirmed also by numerical simulations) demonstrate a remarkable efficiency in the far field, with a reduction up to 15 dB with respect to the absence of the ANC system.     

Adaptive Laguerre filters for active noise control

Feedforward controllers are used in many active noise control (ANC) systems to generate destructive interference in noise fields. An ideal feedforward ANC controller should have an infinite impulse response (IIR) transfer function, but most available feedforward ANC controllers have finite impulse responses (FIR) instead. The main reason is related to the adaptation algorithms of ANC systems. In general, adaptive FIR filters converge faster with guaranteed stability. In this study, the adaptive Laguerre filter is proposed and tested in an ANC application with positive experimental effects. The new ANC controller is an IIR filter, but its adaptation is similar to that of a FIR filter with fast convergence and guaranteed stability. Detailed explanations and analysis are presented in the main text.