Proposals for the directive characterisation of electroacoustic emitters

In this paper, the concept of directivity is generalised to acoustic sources radiating transients or signals evolutive with time. (These are the most common cases in the power or reinforcement electroacoustic systems.) The generalisation proposed is based upon the anamorphism relating the signal levels emitted into the free space in different directions. The relationship between the signals observed in two arbitrary directions is essentially independent of time. Therefore, the anamorphical relationship offers the possibility of obtaining the directivity patterns simply by using as a test signal that signal commonly emitted by the system (i.e. speech in a reinforcement system for a conference room). This principle and method can be applied without major restrictions to any other system, or piece or part of machinery emitting acoustic energy in discontinuous form.Concerning electroacoustic sources, it appears advantageous to replace the usual test signal consisting of pure tones by the signal proper to the system (music, speech, etc.) filtered into the standardised frequency bands. The complete signal (not filtered) can also give significant results. As a simplifying and reasonable compromise regarding the directivity for speech and music, bands of white noise are proposed as test signals.     

Improving the efficiency of an acoustooptic modulator with a two-lobe directivity pattern by correcting the two-frequency electric signal

The feasibility of forming an efficient two-lobe directivity pattern at Bragg acoustooptic diffraction is theoretically substantiated for the first time. It is shown that the efficiency of diffraction into main beams can be increased by mixing the two-frequency signal with additional spectral components that correspond to the closest combination frequencies. Mixing is accomplished using an original electronic circuit designed for polynomial nonlinear correction of the control signal. Bragg diffraction by the corrected signal is studied experimentally in a high-efficiency paratellurite (TeO₂) modulator. Satisfactory agreement with the theory is obtained, and results of applied interest are obtained. In particular, the maximal overall efficiency of diffraction into two main beams is increased from 68 to 79%.     

On the acoustic radiation modes of compact regular polyhedral arrays of independent loudspeakers

Compact spherical loudspeaker arrays can be used to provide control over their directivity pattern. Usually, this is made by adjusting the gains of preprogrammed spatial filters corresponding to a finite set of spherical harmonics, or to the acoustic radiation modes of the loudspeaker array. Unlike the former, the latter are closely related to the radiation efficiency of the source and span the subspace of the directivities it can produce. However, the radiation modes depend on frequency for arbitrary distributions of transducers on the sphere, which yields complex directivity filters. This work focuses on the most common loudspeaker array configurations, those following the regular shape of the Platonic solids. It is shown that the radiation modes of these sources are frequency independent, and simple algebraic expressions are derived for their radiation efficiencies. In addition, since such modes are vibration patterns driven by electrical signals, the transduction mechanism of compact multichannel sources is also investigated, which is an important issue, especially if the transducers interact inside a shared cabinet. For Platonic solid loudspeakers, it is shown that the common enclosure does not lead to directivity filters that depend on frequency.     

A beam interpolation algorithm in digital multi-beam sonar

I.IntroductionThedigita1operationins0narsystembringsagreatadvantageinbeamforming.Itispos-sib1etorealizesomcalg0rithmsindigita1technique,whichisusua11yhardtoorimpossib1etoAnp1ementintraditiona1ana1ogues0nar.Asanexamp1e,ageometrygraphofcirclearrayisil-1ustratedinFig,1.Thetotalnumber0farraye1ementisN.Therearetwokindsofbeam.TheA-c1assbeamconsistsof2M-le1ements,thesteeringdirectionisjustlocatedintheposi-honofoneofthearrayelement;TheB-classbeamconsistsof2Me1ementsandthesteeringdirectionisinthem…     

Precise bearing and automatic target tracking in digital sonar

I.IntroductionItiswellknownthatthepreciseestimationofDOA(DirectionOfArrival)oftargetisanimportantthesisinpassivesonardesign.TheoptimumestimationaccuracyexpressionforlinearrayispresentedbyLewisalldothersll-3].Basedonthepointofviewoftime-delayestimation,Carteretal.showthattheaccuracyoftime-delayestimationofreceivedsignalbetweentwodifferentchannelscanreachtheoptimumCramer-RaolowboundI4-6].Forthedigitalsonarthebearingbeamdirectionisusuallyequal-spaceddistributedwithino-36o".Inordertofindthep…     

A method for estimating the orientation of a directional sound source from source directivity and multi-microphone recordings: principles and application

Taking into account directivity of real sound sources makes it possible to try solving an interesting and biologically relevant problem: estimating the orientation in three-dimensional space of a directional sound source. The source, of known directivity, produces a broadband signal (in the ultrasonic range, in this application) that is recorded by microphones whose position with respect to source is known. An analytical method to process the recorded signals and estimate source orientation is developed in this paper. Experiments testing method performance in estimating source orientation were performed both in a laboratory environment with a Polaroid transducer as source and in a flight room with a Myotis daubentonii bat. In the first case, results showed the estimation method to be accurate and pointed out its limitations. The latter case is significant as an example biological application of the method for extracting behavioral features from bats; results are compared with alternative calculations based on microphone root-mean-square (rms)-pressure values.     

An airborne parametric array

The development of a prototype parametric speaker comprised of 37 piezo-speakers, also known as “acoustic flashlight” is described. The sound pressure along the axis was measured for both the primary (38.5 kHz ultrasonic) signal and the secondary (demodulated) signal. Comparison with the theory for an equivalent piston radiating at the same frequency and the simulation of the piezo speakers as simple sources in the same hexagonal arrangement was made. The absorption coefficient and the radiation patterns of the primary and secondary signal were also measured. All the measurements and the simulation agree with the corresponding theories. The demodulated (audio frequency) signal behaves in the near field of the speaker in the same way (that of an equivalent piston) as the primary (ultrasonic frequency) signal, which results in the inheritance of its high directivity.     

Real time decomposition of speech into modulated components

Motivated by the active process of the outer hair cell (OHC) in the mammalian inner ear, a real time decomposition of speech into modulated components is presented. A generalized phase lock loop (GPLL) was applied to decompose the speech signal into its envelope and positive instantaneous frequency (PIF) parts, which can be further processed and represented by timing information alone. A log-derivative operator is applied to the bandpass signal. Analytic and antianalytic components occupying non overlapping frequency bands are separated by filtering. The proposed algorithms are used to represent speech signals processed through a bandpass filter bank.     

一种高速双光路并行数据采集系统的设计

本文提出了用时分复用方法实现双光路并行信号的模数转换方案,讨论了保证精度和并行性的条件,如A/D转换器精度、信号的频率和孔径误差等,给出了相应的计算公式,在此基础上,介绍了一种高速双光路并行数据采集系统的设计及其应用。在此系统中,用PLD设计的状态机是实时读取数据的关键部分。     

Influence of component errors on the directivity function of pulsed, ultrasonographic linear arrays

The directivity of linear arrays can deteriorate by reason of component errors such as variations in the sensitivity of the array elements, and variations in the electronic delays applied to synthetic aperture focusing. A prototype 16 channel linear array system has been analysed by calculating the directivity functions corresponding to random-error ensembles in relation to sensitivity and delay. With state-of-the-art components, the main lobe of the directivity function is hardly influenced, but the tails of the function show stronger variation, and also become asymmetrical.     

On Complex Delay Time of a Narrow-Band Signal Passing through a Resonant Filter

It is shown that distortion of a narrow-band signal with a gradually varying envelope during passage through a resonant filter can approximately be described in terms of complex delay time. The real part of the delay time can be both positive and negative, depending on the filter type and the carrier frequency of the signal. The imaginary part of the delay time leads to a distortion of the complex envelope of the signal. The intensity envelope of signals without frequency modulation is distorted only slightly, while distortion is considerable even in the case of a fairly small frequency modulation. __________ Translated from Izvestiya Vysshikh Uchebnykh Zavedenii, Radiofizika, Vol. 48, No. 8, pp. 711–718, August 2005.     

A background subtraction approach based on complex wavelet transforms in EDXRF

Subtracting the background accurately is one of the most important issues in energy‐dispersive X‐ray fluorescence (EDXRF) spectra processing. This paper presents a novel approach to perform background subtraction based on dual‐tree complex wavelet transform. Compared with real wavelet transform, the proposed method has some attractive properties, including a smooth, nonoscillating, and nearly shift‐invariant magnitude with a simple near‐linear phase encoding of signal shifts. Therefore, it outperforms real wavelet transform to decompose background into low‐frequency components. The effectiveness of the proposed approach is demonstrated via two simulated spectra with different kinds of backgrounds and one measured spectrum from an energy‐dispersive X‐ray spectrometer. Both simulated and experimental results prove that the proposed approach can subtract background in EDXRF spectra effectively. Copyright © 2014 John Wiley & Sons, Ltd.     

An Improved Car-Following Model for Multiphase Vehicular Traffic Flow and Numerical Tests

This paper attempts to introduce an improved difference model that modifies a car-following model, which takes the next-nearest-neighbor interaction into account. The hnprovement of this model over the previous one lies in that it performs more realistically in the dynamical motion for small delay time. The traffic behavior of the improved model is investigated with analytic and numerical methods with the finding that the new consideration could further stabilize traffic flow. And some simulation tests verify that the proposed model can demonstrate some complex physical features observed recently in real traffic such as the existence of three phases： free flow, coexisting flow, and jam flow; spontaneous formation of density waves; sudden flow drop in flow-density plane; traffic hysteresis in transition between the free and the coexisting flow. Furthermore, th.e improved model also predicts that the stable state to relative density in the coexisting flow is insusceptible to noise.     

Model-based analysis of dispersion curves using chirplets

Time-frequency representations, like the spectrogram or the scalogram, are widely used to characterize dispersive waves. The resulting energy distributions, however, suffer from the uncertainty principle, which complicates the allocation of energy to individual propagation modes (especially when the dispersion curves of these modes are close to each other in the time-frequency domain). This research applies the chirplet as a tool to analyze dispersive wave signals based on a dispersion model. The chirplet transform, a generalization of both the wavelet and the short-time Fourier transform, enables the extraction of components of a signal with a particular instantaneous frequency and group delay. An adaptive algorithm identifies frequency regions for which quantitative statements can be made about an individual mode's energy, and employs chirplets (locally adapted to a dispersion curve model) to extract the (proportional) energy distribution of that single mode from a multimode dispersive wave signal. The effectiveness of this algorithm is demonstrated on a multimode synthetic Lamb wave signal for which the ground-truth energy distribution is known for each mode. Finally, the robustness of this algorithm is demonstrated on real, experimentally measured Lamb wave signals by an adaption of a correlation technique developed in previous research.     

Maximization of the directivity ratio with the desired audible gain level for broadband design of near field loudspeaker arrays

This paper mainly addresses design methods for near field loudspeaker arrays. These methods have been studied recently since they can be used to realize a personal audio space without the use of headphones. From a practical view point, they can also be used to form a directional sound beam within a short distance from the sources especially using a linear loudspeaker array. In this regard, we re-analyzed the previous near field beamforming methods in order to obtain a comprehensive near field beamforming formulation. Broadband directivity control is proposed for multi-objective optimization, which maximizes the directivity with the desired gain, where both the directivity and the gain are commonly used array performance measures. This method of control aims to form a directive sound beam within a short distance while widening the frequency range of the beamforming. Simulation and experimental results demonstrate that broadband directivity control achieves higher directivity and gain over our whole frequency range of interest compared with previous beamforming methods.     

基于实时主动声定位的快速测频算法

针对爆炸试验中冲击波超压传感器位置定位的问题,采用陆上实时主动声定位系统对爆炸试验中冲击波超压传感器位置进行定位。系统中对主动发出声信号频率为已知特定几种频率之一的单频信号的快速实时频率检测,提出了一种快速频率检测算法。算法充分利用信号频率为已知特定几种频率之一这一特定条件,并结合离散时间傅里叶变换(DTFT)与离散傅里叶变换(DFT)的关系以及帕塞瓦尔定理,得出快速频率检测的表达式。经实验验证和计算分析,算法检测频率精准可靠,计算量比其他快速测频算法小得多,更适用于军事紧急状态下陆上实时主动声定位中主动发声和实时定位条件下的快速测频。     

一种基于两块PEM调制器的光谱测量方法

弹光调制干涉信号范围为百赫兹到数十吉赫兹之间,而由于探测器阵列无法对该等级频率实现有效响应,因此,该情况使弹光调制器在光谱成像工作中受到限制。为了解决该问题,发展了一种使用两块具有相近谐振频率的PEM,并基于该频率差进行光信号调制的方法。该方法将两个弹光调制器分别工作在数值略有差异的频率f₁和f₂上,被测光通过双弹光调制器实现差频调制,因此干涉信号中产生载有被测光的低频调制分量,低频调制频率是以δ_i(σ,t)=δ_0i(σ)sin(ω_it)为基频的一系列倍频信号,该低频调制信号使用普通探测器即可实现探测,再将直流和高频信号滤波后,仅对调制信号后的低频成分进行对应的运算即可得到被测光谱。由于该频率差比所使用PEM的谐振频率低2至3个数量级,因此,该方法可使探测器获得更多的响应时间,而且由于该方法并不需要所使用的两块PEM具有严格一致的谐振频率和相同的光程差,降低了系统本身的设计难度。     

A new blind fault component separation algorithm for a single-channel mechanical signal mixture

A vibration signal collected from a complex machine consists of multiple vibration components, which are system responses excited by several sources. This paper reports a new blind component separation (BCS) method for extracting different mechanical fault features. By applying the proposed method, a single-channel mixed signal can be decomposed into two parts: the periodic and transient subsets. The periodic subset is related to the imbalance, misalignment and eccentricity of a machine. The transient subset refers to abnormal impulsive phenomena, such as those caused by localized bearing faults. The proposed method includes two individual strategies to deal with these different characteristics. The first extracts the sub-Gaussian periodic signal by minimizing the kurtosis of the equalized signals. The second detects the super-Gaussian transient signal by minimizing the smoothness index of the equalized signals. Here, the equalized signals are derived by an eigenvector algorithm that is a successful solution to the blind equalization problem. To reduce the computing time needed to select the equalizer length, a simple optimization method is introduced to minimize the kurtosis and smoothness index, respectively. Finally, simulated multiple-fault signals and a real multiple-fault signal collected from an industrial machine are used to validate the proposed method. The results show that the proposed method is able to effectively decompose the multiple-fault vibration mixture into periodic components and random non-stationary transient components. In addition, the equalizer length can be intelligently determined using the proposed method.     

有限振幅声波间的非线性相互作用对声源远场指向性的影响

根据Fenlon理论推导得到了多频声源的辐射声压.基于单频声源谐波指向性的求解方法,得到了二阶近似下的双频声源辐射出的声波相互作用时的远场指向性.分别研究和讨论了在初始辐射声压和频率不同情况下,两列波的相互作用对其中一列波的一阶波和二阶波远场指向性的影响.结果表明,声波间的相互作用对声源远场指向性的影响根据各波之间的相对初始辐射声压和相对频率的不同而有所变化. 关键词： 有限振幅声波 非线性相互作用 远场指向性