Active noise control

Active noise control exploits the long wavelengths associated with low frequency sound. It works on the principle of destructive interference between the sound fields generated by the original primary sound source and that due to other secondary sources, acoustic outputs of which can be controlled. The acoustic objectives of different active noise control systems and the electrical control methodologies that are used to achieve these objectives are examined. The importance of having a clear understanding of the principles behind both the acoustics and the electrical control in order to appreciate the advantages and limitations of active noise control is emphasized. A brief discussion of the physical basis of active sound control that concentrates on three-dimensional sound fields is presented     

The active control of sound

Active control of sound results from destructive interference between the sound field of an original acoustic source and that from a controllable array of `secondary' acoustic sources. For this destructive interference to occur over an appreciable region of space the sound field of the secondary sources must match that from the primary source in both time and space. The spatial matching requirement leads to an upper frequency of applicability of active control. Active control complements conventional passive methods of sound control, which do not work well at low frequencies. Practical feedforward controllers, using a multichannel generalisation of the well known LMS adaptive algorithm, have been developed, using as many as 16 loudspeakers and 32 microphones, and applied with considerable success in the control of low-frequency propeller noise inside aircraft and low-frequency engine noise inside cars. The authors describe such systems     

Autoregressive modeling of lung sounds: characterization of sourceand transmission

In this communication, we discuss the application of autoregressive modeling to lung sounds analysis. The lung sounds source in the airway is modeled as a white noise source, consisting of one or a combination of the following sources: random white noise sequence, periodic train of impulses, and impulsive bursts of energy. The acoustic transmission through the lung parenchyma and chest wall is modeled as an all-pole filter. Using this method, the source and transmission characteristics of lung sounds are estimated separately, based on the lung sounds at the chest wall. To illustrate the potential validity of the model, lung sound segments in known disease conditions were selected from teaching tapes and the source and transmission characteristics were estimated by applying the model. The estimated characteristics were found to be consistent with current knowledge of the generation and transmission of lung sounds in the known conditions.     

声阵列技术在雷达传动系统异响故障诊断中的应用

声阵列技术通过特定方式排布的声传感器阵列接收声场信号,采用波束形成原理对声场信号进行特殊处理,获取噪声源的幅值、相位、方位等信息,得到声场的空间分布,实现混响环境多个声源的检测与识别。针对某轮轨式大型相控阵雷达天线座滚轮的异响噪声故障,基于声阵列噪声源识别技术,利用平面螺旋声像检测系统对滚轮噪声的辐射声场进行测量,获得了滚轮噪声的频率特性和声场能量分布特征,并通过声场图像和光学视频图像的透明重叠,以云图方式准确直观地呈现了滚轮噪声源的频率、分布位置。研究结果表明:声阵列技术能够快速准确地进行噪声源的诊断和空间定位,为雷达传动系统异响噪声故障的诊断分析提供了有效的工程经验。     

基于小型阵列探头的声波分离方法研究

声级计等现有的声压在线测量工具并不能够抑制干扰噪声对测量精度的影响,针对这一问题提出了一种基于小型阵列探头的声波分离方法,该方法以球波函数叠加逼近理论作为分离算法的基础,可应用于噪声环境下目标对象辐射声压的简易快速准确测量。为验证该方法的可行性,设计了一个由5只传声器组成的小型立体阵列探头,在含有干扰声源的声场中,通过对探头所测得的复声压信号进行声波分离算法处理,便可将测量对象辐射的声波从存在干扰声波的声场中分离开来。数值仿真及实验结果表明,通过该方法可以简易快速地得到较好的声波分离效果。     

Localizing Lung Sounds: Eigen Basis Decomposition for Localizing Sources Within a Circular Array of Sensors

Lung disorders or injury can result in changes in the production of lung sounds both spectrally and regionally. Localizing these lung sounds can provide information to the extent and location of the disorder. Difference in arrival times at a set of sensors and triangulation were previously proposed for acoustic imaging of the chest. We propose two algorithms for acoustic imaging using a set of eigen basis functions of the Helmholtz wave equation. These algorithms remove the sensor location contribution from the multi sensor recordings using either an orthogonality property or a least squares based estimation after which a spatial minimum variance (MV) spectrum is applied to estimate the source locations. The use of these eigen basis functions allows possible extension to a lung sound model consisting of layered cylindrical media. Theoretical analysis of the relationship of resolution to frequency and noise power was derived and simulations verified the results obtained. Further, a Nyquist’s criteria for localizing sources within a circular array shows that the radius of region where sources can be localized is inversely proportional to the frequency of sound.The resolution analysis and modified Nyquist criteria can be used for determining the number of sensors required at a given noise level, for a required resolution, frequency range, and radius of region for which sources need to be localized.     

低噪声电子听诊器的设计

心音、呼吸音是人体重要的声音信号,是医生进行心脏和呼吸系统疾病诊断的重要信息依据。而一般的听诊器引入的噪声较大,影响了诊断的正确性。为实现提高听诊的准确度,需要设计一种低噪声电子听诊器电路。此设计主要包括低噪声的前置级放大电路、滤波电路以及功率放大电路。通过这种低噪声电子听诊器的电路设计,实现了对心音和呼吸音的低噪提取,为实现对心音和呼吸音的正确诊断奠定了良好的基础。     

Design, construction, and evaluation of a bioacoustic transducer testing (BATT) system for respiratory sounds

Many different transducers are employed for recording respiratory sounds including accelerometers and microphones in couplers. However, there is no standard lung sound transducer or any device to compare transducers so that measurements from different laboratories can be determined to be of physiologic origin rather than technical artifacts of the transducers. To address this problem, we designed and constructed a prototype of a device that can be used to compare accelerometers, microphones enclosed in couplers, and stethoscopes. The prototype device consists of a rigid chamber containing a loudspeaker that opens to an antechamber covered by a viscoelastic material with mechanical properties similar to human skin and subcutaneous tissue. When driven by a white noise source, we found the sound output at the surface to be useful to comparatively evaluate sensors between 100 and 1200 Hz where lung sounds have most of their spectral energy. We compared the viscoelastic layer to similar thicknesses of fresh meat and fat and found them to produce similar acoustic spectra. This device allows air-coupled transducers, accelerometers, and stethoscopes used in respiratory sounds measurements to be compared under physical conditions similar to their intended use.     

头戴式有源噪声控制系统的分区域优化控制方法研究

本文提出了一种适用于头戴式有源噪声控制系统的有源降噪方法,结合虚拟传声器技术,并通过将水平面划分为多个子区域,实现人耳处对不同方向噪声的有效控制。初级声场传递函数的实际值与其估计值的偏差是影响头戴式有源噪声控制系统降噪量的主要因素,因此需要优化一定角度范围内的初级声场传递函数,本文提出一种基于最大化该区域最小降噪量的优化设计方法。最后,将水平面划分为不同子区域,每个子区域分别使用对应的优化初级声场传递函数进行噪声控制。实验结果验证了该噪声控制方法的有效性。      

Acoustic FMRI noise: linear time-invariant system model

Functional magnetic resonance imaging (fMRI) enables sites of brain activation to be localized in human subjects. For auditory system studies, however, the acoustic noise generated by the scanner tends to interfere with the assessments of this activation. Understanding and modeling fMRI acoustic noise is a useful step to its reduction. To study acoustic noise, the MR scanner is modeled as a linear electroacoustical system generating sound pressure signals proportional to the time derivative of the input gradient currents. The transfer function of one MR scanner is determined for two different input specifications: 1) by using the gradient waveform calculated by the scanner software and 2) by using a recording of the gradient current. Up to 4 kHz, the first method is shown as reliable as the second one, and its use is encouraged when direct measurements of gradient currents are not possible. Additionally, the linear order and average damping properties of the gradient coil system are determined by impulse response analysis. Since fMRI is often based on echo planar imaging (EPI) sequences, a useful validation of the transfer function prediction ability can be obtained by calculating the acoustic output for the EPI sequence. We found a predicted sound pressure level (SPL) for the EPI sequence of 104 dB SPL compared to a measured value of 102 dB SPL. As yet, the predicted EPI pressure waveform shows similarity as well as some differences with the directly measured EPI pressure waveform.      

基于fluent的高速列车受电弓主被动整体降噪研究

针对高速列车气动噪声越来越大的问题,本文以高速列车某车型为参考建立1:1受电弓区域局部模型,基于宽频带噪声源模型、LES大涡模拟及FW-H声学模型,运用弓头仿生降噪和底部空腔主动射流降噪的整体降噪措施,采用数值模拟法研究高速列车受电弓区域的降噪效果。结果表明:受电弓弓头和底部空腔是气动噪声的主要来源;降噪后,主要噪声源的声功率级都有了较大降幅,其中弓头和空腔部位分别降低了15.28 d B和16.92 d B;中高楼层住宅处的降噪效果更佳,最大声压级降低位置在距地面18 m高处(距受电弓25 m远处),降低了4.94 d BA;远场声压级在低频区域降噪效果更为显著,特别是在800 Hz位置声压级降幅最大,降低了8.21 d BA。     

运动声源定位阵列设计及校准研究

在飞跃噪声源定位测量中,核心技术是运动声源定位技术,测量过程涉及传声器阵列设计技术及传声器阵列校准基础技术研究.利用对消声室内移测架携带的不同速度的运动声源进行定位分析,利用自研程序处理采集数据获得噪声源实际位置,通过结果对比分析,验证了运动声源定位的可行性,为飞机飞行噪声源定位奠定了基础.     

Hybrid excitation of SRM for reduction of vibration and acoustic noise

The inherent simplicity, ruggedness, and low cost of a switched reluctance motor (SRM) makes it a viable candidate for various general-purpose adjustable-speed applications such as industrial and home appliances. The primary disadvantage of an SRM is the vibration and acoustic noise compared with that of conventional machines. The main source of vibration in the SR drive is generated by rapid change of radial magnetic force when phase current is extinguished during commutation action. In this paper, a hybrid excitation method with C-dump inverter is proposed to reduce vibration and acoustic noise in the SR drive. The hybrid excitation has two-phase excitation by long dwell angle as well as conventional one-phase excitation. The vibration and acoustic noise are reduced because the scheme reduces abrupt change of excitation level by distributed and balanced excitation.     

A novel prediction method of acoustic magnetic noise based oninduction motor's NHCC function

When an induction motor is driven by a pulsewidth modulation (PWM) inverter, acoustic magnetic noise caused by harmonics of the induction motor input is a serious environmental problem. In this paper, a novel prediction method of acoustic magnetic noise is proposed. Acoustic magnetic noise emitted from an induction motor is considered as the function of harmonic components of induction motor current. The acoustic magnetic noise versus harmonic current characteristic (NHCC) function is defined. This NHCC function describes the response of acoustic magnetic noise to a single-frequency harmonic current. Based on the NHCC function, acoustic magnetic noise can be predicted very easily and accurately, simply by the calculation of harmonic components of the induction motor current. By making use of this method, the difficulties of the conventional prediction method, such as the complex calculations of magnetic force, mechanical resonant frequencies, and sound radiation efficiency can be avoided. This novel method can be used as a tool for environmental noise management and evaluation of low-acoustic magnetic noise characteristics of different inverter PWM control methods     

Joint effect of artificial noise and primary interference on security performance of cognitive radio networks

Underlay mechanism allows concurrent communications of primary users and secondary users in cognitive radio networks (CRNs), causing mutual interference between them. However, current literature neglects primary interference or considers it as Gaussian noise. In addition, artificial noise, which is intentionally generated to interfere eavesdroppers, can improve security performance of CRNs. This paper analyzes security performance of CRNs, accounting for artificial noise and considering primary interference as non-Gaussian noise, under maximum transmit power constraint, interference power constraint, and Rayleigh fading channels. The security performance is evaluated through proposed exact expressions of secrecy outage probability, non-zero achievable secrecy rate probability, and intercept probability, which are verified by Monte-Carlo simulations. Various results demonstrate that CRNs suffer security performance saturation in the range of large maximum transmit power or large maximum interference power, and primary interference significantly deteriorates security performance while artificial noise is useful in enhancing this performance.     

基于舰船辐射噪声的舰船目标定位技术

通过声传感器阵列可实现对声源的自适应检测定位及跟踪,这使得它在目标探测、故障诊断等诸多领域有着广泛的应用。在复杂环境中,常规的相关方法对于声源的定位往往难以得到准确的定位结果。针对常规时差测向定位抗干扰性能差和易受噪声影响,特别是噪声存在显著的尖峰时更不能得到满意结果的问题,本文提出基于盲分离的定位算法改进其性能。通过仿真实验,可以看到本文提出的定位方法能够取得更稳定、准确的结果。     

Combined acoustic echo cancellation,dereverberation and noise reduction: a two microphone approach

This paper presents new algorithms for acoustic echo cancellation and noise reduction which use two (or possibly more) microphone signals. In contrast to the single microphone method the multimicrophone approach can exploit the spatial coherence properties of sound fields which arise from noise and reverberated speech. Besides the standardfir echo canceller the proposed algorithms comprise an adaptive filter to eliminate non coherent signal components. The combined system achieves better Erle than thefir echo canceller alone, attenuates ambient noise, dereverberates near end speech, and possibly leads to implementations with reduced complexity. The paper analyzes the acoustical properties of typical environments, presents the algorithms and experimental results.     

利用相位时频掩蔽的麦克风阵列噪声消除方法

本文提出了一种在干扰声源和背景噪声存在条件下麦克风阵列噪声消除的方法。麦克风阵列通过波束形成增强由导向矢量所指定方向的目标声源来抑制背景噪声。然而,现有的波束形成算法在干扰声源存在的情况下,无法进行准确的导向矢量估计。为此,本文提出一种基于音频信号互相关功率谱相位的麦克风阵列噪声消除方法。首先通过音频信号的相位时频掩码估计导向矢量,并对其进行波束形成,从而有效抑制干扰声源和背景噪声;然后利用语音存在概率,采用最大似然的方法估计波束形成后信号中残留的干扰噪声功率谱密度,对其进行后处理,进一步抑制残留干扰和噪声。实验结果表明在干扰声源和背景噪声存在的条件下,所提方法有效地实现了麦克风阵列噪声消除,且各种性能指标优于基线方法。      

低频水声通信系统水听器流噪声数值分析

低频水声通信是实现远距离水下无线通信的一种重要方式，在海洋科考、环境监测及军事等领域应用广泛。流噪声是制约低频水声通信传输距离的重要因素之一。不同于以往对充油的拖曳线列阵声呐的流噪声研究，由于低频水声通信系统常用换能器和裸露水听器实现水下信号的发送和接收，本文以该系统中裸露水听器流噪声为研究对象，采用大涡模拟方法（Large Eddy Simulation, LES）和声学类比方法方法对水听器周围的流场、声场进行数值模拟计算分析，研究在不同外形参数、运动参数下的水听器周围的流噪声。分析对比结果表明裸露水听器周围的流噪声对低频水声通信存在影响，且流噪声声压级随着运动速度的增加而提高。本研究为提升远程低频水声通信性能提供了噪声方面的理论基础。      

基于光纤光栅传感器的心音信号提取与处理算法研究

本文研究了一种用于光纤光栅智能服装的心音信号提取与处理算法,实现异常心音的初步识别。提出基于希尔伯特-黄变换（HHT）和小波阈值消噪相结合的心音提取算法,对波长解调信号进行消噪,提取有用的心音信号。采用数学形态学进行心音包络提取,提出基于直线结构元素和余弦结构元素相结合的心音处理算法,准确获取心音峰值点和起止点位置并计算心音特征值,根据心音特征值的临床意义,判断心音是否正常。实验结果表明该算法能够有效消除波长解调信号中的呼吸干扰与噪声,对20例实测正常心音和8类常见异常心音均能正确识别。该算法具有易实现、识别率高的特点,对光纤传感智能服装的研发和心脏疾病的早期诊断具有重要意义。