调制气流声源的原理

利用气流调制以产生声音是一个比较有效的方法。人的发声器官、旋笛、电动气流扬声器以及一般哨子等都是根据这个原理。气流声源的特点是效率高,功率可以很大,有实用价值。关于气流声源的发声原理,过去已有一些讨论,但有的是过分简化,有的比较细致,但又过于繁复,不便计算。本文从气体动力学的基本原理出发,求得气流声源的气流-压力特性,并用图解法求得在给定气室压力和气流喷口面积比的条件下声辐射特性的方法,考虑到气流声源的气流特性和辐射关系都是非线性的,但是辐射声功率主要由基频决定,从而用近似理论求得了声功率、气流产额和气流效率的简单表达式,和严格理论的结果相比,误差不到1分贝。文中把结果画成图表以便计算,并讨论了最佳设计的问题。     

选择声强法用于强背景噪声下声源识别

选择声强法是一种新的测试方法,它集中了传统声强法和偏相干方法的优点,可以在声源间存在相干和高背景噪声的情况下,进行声源辐射声强的识别.本文首先论述了选择声强法的理论,并在多输入、双输出的频域模型基础上,讨论了如何利用选择声强法将声强矢量分解为与各噪声源有关的分量,然后结合在强背景噪声下扬声器辐射声强的区分实验,给出了此方法在实际应用中可能遇到的一些问题以及相应的解决办法.     

大功率调制气流声源阵列的相干合成实验研究

完成了调制气流声源阵列的相干合成实验. 提出了利用主动相位控制方法实现调制气流声源阵列相干合成的思路, 介绍了基于随机并行梯度下降算法的声源阵列相干合成的原理. 对利用该算法实现声源阵列的相干合成进行了数值模拟, 完成了双调制气流声源阵列在远场的相干合成实验, 并给出算法参数的合理设置方案. 实验结果显示, 基频成分的相干合成效果明显, 算法收敛时测点处的声压级相比单源发射增加了4 dB, 接近于各单源功率谱中基频成分相干合成、其他频率成分非相干合成的结果; 结果表明实验中算法能够有效控制各调制气流声源辐射声波的相位, 取得了明显的相干合成效果. 关键词： 调制气流声源 相干合成 随机并行梯度下降 高阶谐波     

光电检测方法在大功率气流声源实验研究中的应用

内流场和调制面积的准确测量是大功率调制气流声源实验研究中的关键问题,传统测量技术具有明显的局限性。采用光电检测方法以实现速度场和声源调制面积的非接触测量,通过粒子速度成像技术获取声源气路系统中的速度场,通过CCD系统实现对声源振动系统位移输出和调制面积变化的实时监控。实验结果与数值模型结果的比较验证了测试方法的可行性。速度场结果表明,典型工况下受限射流靠近内侧壁面,流动分离出现在外侧壁面附近,声源内部形成的剪切层位于主流与回流之间。音圈位移实测结果说明所选择的驱动参数可实现对喷口射流的满调制。     

调制声源的近场声全息分析

为了解决传统近场声全息技术在重建调制声源声场时无法显示出调制成分的问题,先通过Hilbert变换对调制信号的声场进行解调处理,将调制信号和载波信号分离开来,然后再对调制成分进行普通的重建分析,由此提出了基于Hilbert变换的调制声源近场声全息分析方法.文中的数值仿真与试验研究均表明此方法可以有效地完成调制声源的重建分...     

低频声波油井液面检测方法研究

为克服冲击声波法油井液面检测设备寿命短的缺陷,研究了一种低频声波测量油井内液面深度的方法,采用电磁式扬声器产生伪随机码调制的低频声波作为探测信号。通过对回波信号的解码和相关检测处理可以抑制背景噪声的干扰。完成了油井液面检测仪的硬件和软件系统的初样设计,并对不同信噪比下的回波信号作了仿真分析,仿真分析结果表明,该方法在低信噪比情况下,仍具有良好的辨识效果。     

“三点法”测量强声源声功率

目前强声源的输出声功率达几千瓦甚至几万瓦,它们多数是气流声源,因此用通常的方法测量它们的声功率会遇到困难,例如对于行波管法,终端吸声处理比较困难,而当终端反射系数达0.25时,声场起伏有2.5dB,测量误差较大。本文利用声在管中传播的基本原理,提出由管中纵向分布的三点声压确定声源声功率的方法,这个方法允许管终端存在反射。误差分析指出,终端反射系数为0.25时,声场起伏引起的测量误差小于0.2dB,对两千声瓦电动调制气流扬声器的实际测量表明,当反射系数高达0.6时,多次测量(各次测量时管中纵向分布的三点位置各不相同)结果比较一致,偏差小于0.6dB。证实这种方法具有设备简单,结果准确的优点。     

微通道板输入信号利用率提高研究

分析了微通道板输入信号损失的原因,提出了在微通道板输入端镀制绝缘层,从而提高微通道板输入信号利用率的方法,并进行了试验.试验结果表明:在微通道板输入端镀制一层15nm的绝缘层,可以提高微通道板输入信号的利用率,从而提高微通道板的增益.绝缘层的二次电子发射系数越高,微通道板输入信号的利用率越高,增益提高的比例越大.对SiO2膜层而言,可以提高12%左右;对Al2O3膜层而言,可以提高35%左右.在微通道板增益提高的同时,像增强器的分辨力和调制传递函数会降低,并且绝缘层的二次电子发射系数越高,分辨力和调制传递函数降低的比例越大.但微通道板分辨力和调制传递函数降低的比例远低于增益提高的比例.本文提出的提高微通道板输入信号利用率的方法具有一定的实用性,可以推广使用.     

一种简便的表达声波波动特性的演示方法

1引言 普通扬声器发出的声波不是强指向性的；当声源发出的直达声波到达接收传声器后，由房间内对应各个反射面向声源产生的声波也陆续到达，造成严重的干扰．由于室内反射面状况较复杂，对时域中具有一定持续时间的信号，其直达声与反射声完全混叠，无法将各个像声源所对应的声波一一辨别出，因此直接采用扬声器发出的窄带或单频信号作为波源的实验方法就无法应用于普通的课堂教学．     

面向目标的定向声辐射技术及其应用研究

面向目标的定向声辐射是指利用声源特性和阵列信号处理技术,将声波传送至目标方向或区域的声场控制方法。本文介绍了定向声辐射技术的原理和进展,重点阐述了普通扬声器阵列和参量阵扬声器这两种声源形式的理论、算法和应用研究。其中,针对扬声器阵列,主要分析了基于声能量对比度控制的声聚焦技术;针对参量阵扬声器,综述了利用非线性效应产生高指向性可听声的计算方法、信号预处理和应用等进展。面向目标的定向声辐射技术的研究对实现智能声场控制有着重要意义。     

On preprocessing techniques for bandlimited parametric loudspeakers

The self-demodulation property of finite-amplitude ultrasonic waves can be applied with parametric loudspeaker to produce audible sound. A special characteristic of the reproduced sound waves using parametric loudspeaker is its high directivity. However, the demodulated signal from parametric loudspeaker suffers from high distortion. To reduce the distortion in the demodulated signal, preprocessing of the modulating signal is usually employed. To determine the effectiveness of the preprocessing technique, an important practical constraint on the bandwidth of the ultrasonic transducer of the parametric loudspeaker should be accounted. In this paper, we shall discuss a class of preprocessing techniques that is based on quadrature amplitude modulation. As compared to the conventional preprocessing methods used with bandlimited ultrasonic transducer, the demodulated signal from our proposed preprocessing techniques exhibits lower distortion.     

Acoustic characteristics of an electrodynamic planar digital loudspeaker

In this paper, an electrodynamic planar loudspeaker driven by a digital signal is experimentally discussed. The digital loudspeaker consists of 22 voice coils, 11 permanent magnets, a diaphragm with streamlined sections molded in plastic, and a suspension made of handmade Japanese paper between the diaphragm and the frame. First, the acoustic responses are affected by the arrangement of the voice coils, so an asymmetric arrangement is studied. This asymmetric arrangement is designed to obtain as flat a frequency response to an analog signal as possible. This arrangement is compared with a symmetric one and results show that the flatness of the frequency response around 1 kHz and 4 kHz is improved and that the sound reproduction band is from 40 Hz to 10 kHz. Second, to evaluate the acoustic responses to a digital signal, the digital loudspeaker is driven with a pulse code modulation signal. Results show that the digital loudspeaker can reproduce pure sound with a total harmonic distortion of less than 5% from 40 Hz to 10 kHz, exceeding this value only in a narrow frequency band near 4 kHz. This digital loudspeaker was demonstrated to have good linearity over its dynamic range of 84 dB.     

声速测量及声波干涉演示仪器的研制

本介绍了声波干涉、反射的两种物理演示仪器的研制,一种是通过调节波导管中液面的高度形成驻波,实现对声速的测量;另一种是由一个信号发生器驱动两个扬声器,这两个相干的声源在自由空间中形成强弱变化的声波干涉。本分别介绍了这两种装置的原理、制作及使用效果。     

Model and estimation method for predicting the sound radiated by a horn loudspeaker - With application to a car horn

This paper deals with a new car horn device made of a sound synthesizer and an electrodynamic horn loudspeaker. It presents an one-dimensional model allowing to predict the loudspeaker efficiency and a specific method to estimate experimentally the model parameters. First, this model aims at reducing the time spent in the design process. Second it aims at correcting the sound emitted by the sound synthesizer in order that the listener hears the sound designed for creating the warning message. The study gives a survey of the vast loudspeaker literature. It is based on the conventional electroacoustic approach used for electrodynamic loudspeakers and on wave propagation models used for characterizing acoustic horns. The estimation of the model parameter values is performed using measurements of the electrical impedance of the loudspeaker and of the acoustic impedance of the horn. The model is assessed by comparing the calculated and measured electrical impedances and horn efficiencies. Results show that the model predicts well the horn efficiency up to 2500 Hz, the limitation being due to the horn radiation impedance modelization.     

两扬声器虚拟声重放的双耳声压控制与定位性能

在两扬声器虚拟声重放中,通过精确重构双耳声压而产生不同的空间听觉感知。其重放的定位性能应该是由双耳声压控制的代价和稳定性所共同决定的。过去研究主要对双耳声压控制的稳定性进行分析,并以此作为扬声器布置和信号处理的依据。该文研究表明仅对双耳声压的稳定性分析是不足以完全衡量扬声器虚拟声重放的定位性能的。进一步采用虚拟声信号处理滤波器响应平均功率对双耳声压控制的代价进行分析。结果表明,缩窄左右对称扬声器布置的张角或采用非对称扬声器布置会明显增加产生侧向目标虚拟源时的双耳声压控制代价。虚拟源(虚拟声像)定位实验表明,双耳声压控制代价增加会引起虚拟源定位缺陷。实际应用中,为了有效产生侧向虚拟源,应避免采用过窄张角(如立体声偶极)和非对称的扬声器布置。     

金属杆内疲劳裂纹与振动/超声相互作用及其定量表征

根据同步解调的原理,考察振动声调制检测输出信号中的调制成分,对金属杆内振动/超声在裂纹面上的相互作用进行了分析.制作了含有不同尺寸裂纹的铝杆试样,以扫频激励的方式得到合适的高频激励参数,分析输出信号中的调制信息,发现工件中的高频超声可分为两部分:一部分声波通过裂纹面,受到低频振动信号的调制,另一部分则保持不变。在此基础上对非线性调制模型进行了修正,并提出了一种用于裂纹检测的定量方法。与超声C扫描检测结果对比表明,此定量方法可用于估计金属杆中疲劳裂纹尺寸。      

Interaction of fatigue crack with vibration and ultrasound in metallic rod and its use for quantitative characterization

According to the principle of synchronous demodulation,the modulation information in the output signal of vibro-acoustic modulation test was investigated and the nonlinear interaction of sound and vibration with the crack interface in a metallic rod was analyzed.A swept signal excitation was used to obtain a proper high frequency parameter in the test of aluminum rod samples with different sizes of cracks and the modulation information in the output signal was analyzed.It was found that the ultrasound wave in the structure can be separated into two parts.One is the wave which passes through the crack interface modulated by the low frequency vibration,and other wave has no any changes.The nonlinear modulation model was modified and a quantitative method for crack detection was proposed.It is demonstrated by comparing with the ultrasonic C-scanning that this method can be used to estimate the fatigue crack size in the metallic rod.     

On the influence of interaural differences on temporal perception of noise bursts of different durations

The perception of a composite sound's temporal cues, like synchronous onsets, is considered essential to correct perceptual grouping of its constituent components. The processing of a single sound's spatial cues, already present at its onset, may interact with temporal perception of the onset. The current study investigated the influence of interaural differences on temporal perception of a sound's onset. As a measure of temporal perception, the ability to position the onset of a temporally displaced target sound to the regular meter of diotic reference marker sound onsets was measured for various target sound lateralizations, sensation levels, and target and marker sound durations. For target sounds presented in quiet, no influence of interaural differences on temporal positioning of the onset was found. However, increasing a sound's duration systematically shifted the perceived onset position into its "interior." For target sounds presented at low sensation levels in a noise masker, the precision of temporally positioning the onset generally degraded, though faster for dichotic conditions and for longer durations. The level below which temporal perception precision starts to degrade was found to depend on signal-to-noise ratio rather than on sensation level or duration, and is influenced by the presence of interaural differences.     

Sound localization of stereo reproduction with parametric loudspeakers

A parametric loudspeaker radiates an audible signal by the interaction of the primary wave that is amplitude modulated and is known as a super-directivity loudspeaker. The parametric loudspeaker is one of the prominent applications of nonlinear acoustics. So far, the applications have been limited monaural reproduction sound system for public address in museum, station, street etc. In this paper, we investigated sound localization of stereo reproduction using two parametric loudspeakers in comparison with that using two ordinary dynamic loudspeakers. In subjective tests, the binaural information ILD (Interaural Level Difference) or ITD (Interaural Time Delay) was focused on. To investigate the characteristics of sound localization in a wide listening area, three typical listening positions were picked up. Signals were 500 Hz, 1 kHz, 2 kHz and 4 kHz pure tones and pink noise. The used parametric loudspeaker was an equilateral hexagon. The subjective test led to the results that when the parametric loudspeakers were used, the listeners at the three typical listening positions perceived the correct sound localization of not only pure tone but also pink noise and when the ordinary dynamic loudspeakers were used, except for the case of pure tone with ITD, the tendency was almost similar to those using the parametric loudspeakers. The second subjective tests were conducted in order to investigate in details the difference between parametric loudspeakers and ordinary dynamic loudspeakers by increasing the number of subjects. In the case of ITD and 500 Hz using the ordinary dynamic loudspeakers, three types of sound localization were categorized, in which the reversed type was major and the normal and the other types were minor. The ILDs which were measured with a dummy head and were calculated with several formulas were almost the same and indicated the reasons of the reversed typed sound localization and a serious influence of the crosstalk. It was found that in the case of pure tone with ITD, the contradiction between the binaural information ILD and ITD is remarkable, because the directivity of the ordinary dynamic loudspeakers was so dull that the crosstalk components had a serious influence on sound localization. It was determined the parametric loudspeaker could transmit correct binaural information to the listener, because the directivity of the parametric loudspeakers was so sharp that it suppressed the cross talk components.