An alternative method to measure the on-axis difference-frequency sound in a parametric loudspeaker without using an acoustic filter

The self-demodulation characteristic of finite-amplitude ultrasonic sound waves can be applied with parametric loudspeaker to reproduce audible sound with highly directivity. But measuring the difference-frequency sound is still a problem due to the spurious sound generated as a result of nonlinearity caused by the product of the primary waves at the receiving system. In this paper, based on the phase-cancellation method and the Gaussian beam expansion technique, an alternative method is proposed to measure the on-axis difference-frequency sound accurately without using any traditional acoustic filter, where the spurious sound can be greatly reduced or even eliminated. The proposed method is more suitable for the case where the piston source in the parametric loudspeaker comprises multiple small piezoelectric transducers (PZTs) and each transducer element in the array may have different frequency response. The validity of the proposed method is confirmed both by simulations and experiments.     

Identification of a parametric loudspeaker system using an adaptive Volterra filter

Due to the parametric acoustic array effect in air, the input audible signal of a parametric loudspeaker system can be reproduced with high directivity at the target region. However, the reproduced audible signal suffers from harmonic distortion, which is the by-product of nonlinear interaction between the primary waves. In order to investigate this inherent nonlinear phenomenon, a nonlinear system identification model is developed based on an adaptive Volterra filter. Unlike the conventional loudspeaker, the nonlinear characteristic of a parametric loudspeaker system is dependent on several primary parameters in nonlinear acoustics, which include the initial pressure of the primary waves, the observing distance and angle, as well as ambient temperature and relative humidity. By using a truncated Volterra series up to the 2nd-order kernel, numerical simulations are conducted to develop a system model with one group of parameters and examine the quadratic nonlinear intensity for different parameters’ settings. Experimental measurements, which take into account of emitter’s response, are carried out to verify the modeling result and evaluate the model performance. Based on the Volterra system model, the sound pressure level and the harmonic distortion can be accurately predicted.     

Effect of the ultrasonic emitter on the distortion performance of the parametric array loudspeaker

The parametric array loudspeaker (PAL) is a type of directional loudspeaker that utilizes the nonlinear acoustic effect to create the audible sound in an ultrasonic beam. Due to this unusual sound principle, it is inevitable that nonlinear distortion is incurred in the sound transmission of the PAL. Numerous modulation methods aiming to reduce the nonlinear distortion have been developed on the basis of the Berktay’s far-field solution, but they often perform in an unexpected manner. The degraded practical performance has been credited to the inaccuracy of the Berktay’s far-field solution. In this paper, we demonstrate the effect of the ultrasonic emitter on the distortion performance of the PAL and suggest that the Berktay’s far-field solution remains to be a good model equation.     

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。     

FPGA-based implementation of steerable parametric loudspeaker using fractional delay filter

Parametric loudspeaker enables high directivity sound along the radiation direction to be generated by the nonlinear effects of ultrasonic waves in air. Based on the theory of phased array, the beams of parametric loudspeakers can be steered to other directions. In this paper, the Fractional Delay (FD) filter is introduced to generate any arbitrary delay to accurately control the steering angle of the beam. Field Programmable Gate Array (FPGA) is a better choice for the system of steerable parametric loudspeaker due to its strong capability of parallel and real-time processing. The algorithms have been realized easily and quickly by combining the tools of Matlab simulink and DSP Builder. Finally, the validity of the system is verified through comparing the theoretical simulations with the experimental results.     

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.     

扬声器的特性不匹配对虚拟声像的影响

本文从理论和实验上探讨了扬声器的特性不匹配对重放虚拟声像的影响。结果表明,用两扬声器进行虚拟听觉重放时,在某些频率段和虚拟声像角度,两扬声器间很小的幅频特性差异或相频特性差异都足以对虚拟声像方向产生明显的影响。扬声器特性的差异对前方范围的声像影响较小,但对侧向范围的声像影响较大,因而两扬声器的特性不匹配也是导致虚拟听觉重放时侧向声像位置畸变的重要原因之一。而在实际应用中要特别注意两扬声器的特性匹配,或者要用信号处理的方法对两扬声器的特性进行校正。     

Study of vertical sound image control with parametric loudspeakers

A parametric loudspeaker utilizes nonlinearity of a medium and is known as a super-directive loudspeaker. In this paper, the sound localization in the vertical direction using the upper and lower parametric loudspeakers is confirmed by listening tests and physical measurements. The differences in levels between the upper and lower parametric loudspeakers are varied as a parameter. The direction of sound localization in the vertical plane can be controlled not only when the acoustical axis is set to the right ear but also when it is set to at 5 deg to the right of the right ear. The effect of the level difference between the upper and lower loudspeakers is weaker than the differences observed when using ordinary loudspeakers. We obtained interesting characteristics of the left-right sound localization in the horizontal plane with the upper and lower parametric loudspeakers in the vertical plane. It is found that by setting the parametric loudspeaker at the right ear (that is, the horizontal angle of a listener to it is only 3 deg to the right), the direction of sound localization in the horizontal plane moved approximately 10 deg to the right. Moreover, by setting the parametric loudspeaker 5 deg to the right, the direction of sound localization moves approximately 20 deg to the right. The ILD (Interaural Level Difference) using a dummy head is calculated from the measured left and right sound signals. It is determined that ILDs of the parametric loudspeaker are larger than those of the ordinary loudspeaker. A simple geometrical acoustic model is introduced and analyzed. The analysis helps to explain the measured characteristics.     

A new type of high power composite ultrasonic transducer

A new type of high power composite ultrasonic transducer was proposed and studied. The composite transducer consists of a sandwich longitudinal piezoelectric transducer, an isotropic metal hollow cylinder with large radial dimension, and the front and back metal radiation mass. By means of its special structure and Poisson’s effect, the composite transducer can produce vibrations both in its longitudinal and radial directions, and therefore, it can radiate sound waves in three-dimensional space. The electro-mechanical equivalent circuit of the composite transducer was derived and the resonance frequency equation was obtained analytically. Numerical methods were used to simulate the vibration of the composite transducer, and the vibrational displacement distribution, the resonance frequency and the radiation sound field are given. Some composite transducers are designed and manufactured; their resonance frequencies and the radiation acoustic field are measured and compared with the analytical and numerical results. It can be seen that the measured frequencies and acoustic field contour are in good agreement with the analytical and numerical results. It is expected that this kind of composite ultrasonic transducer can be used in more and more power ultrasonic applications, such as ultrasonic cleaning, ultrasonic extraction, ultrasonic sonochemistry and other ultrasonic liquid processing, where high ultrasonic power and large ultrasonic processing space are needed increasingly.     

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.     

头部尺寸对虚拟声像定位的影响

本利用立体声像的相关理论分析方法，分析了用扬声器重发虚拟声时，倾听的头部尺寸对声像定位的影响。结果表明,当实际倾听的头部尺寸与虚拟声信号处理所用头部模型的尺寸有差别时，前方范围内的声像位置畸变较小，但侧向的声像位置畸变较大，因而本特别指出，倾听头部尺寸的不同是虚拟声重发时侧向的声像位置畸变的重要原因，声像定位实验证实了理论分析。     

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.     

复频聚焦超声换能器水中焦区商场特性的研究

本文对新研制的一种复频聚焦超声换能器的辐射声场进行了实验研究,测定了辐射压在轴向上的分布曲线,进而确定了焦区位置,这与理论所得结果符合较好,测定了辐射声压在焦平面上的分布曲线,并对焦区声场进行了定位,最后对换能器在水中焦区辐射声的频谱进行了研究,不仅观测到了两个源波,而且还观测到了和频波,差频波以及倍频波,证实了声散射声效应的存在。     

Wide-band dereverberation method based on multichannel linear prediction using prewhitening filter

Several dereverberation algorithms have been studied. The sampling frequencies used in conventional studies are typically 8–16 kHz because their main purpose is preprocessing for improving the intelligibility of speech communication and articulation for automatic speech recognition. However, in next-generation communication systems, techniques to analyze and reproduce not only semantic information of sound but also more high-definition components such as spatial information and directivity will be increasingly necessary. To decompose these sound field characteristics with high definition, a dereverberation algorithm that is useful at high sampling frequencies is an important technique to process sound that includes high-frequency spectra such as musical sounds. The LInear-predictive Multichannel Equalization (LIME) algorithm is a promising dereverberation method. Using the LIME algorithm, however, a dereverberation signal cannot be solved at high sampling frequencies when the source signal is colored, such as in the case of speech and sound of musical signals. Because the rank of the correlation matrix calculated from such a colored signal is not full, the characteristic polynomial cannot be calculated precisely. To alleviate this problem, we propose preprocessing of all input signals with filters to whiten their spectra so that this algorithm can function for colored signals at high sampling frequencies.     

Reflection and transmission at normal incidence onto air-saturated porous materials and direct measurements based on parametric demodulated ultrasonic waves

The present work is related to the characterization of air-saturated porous media by using parametric demodulated ultrasonic waves. One uses two different powerful ultrasonic emitters working either at 47 kHz or at 162 kHz which are electronically amplitude modulated over the 200 Hz-4 kHz or 2 kHz-40 kHz bandwidths respectively. The demodulation process takes place in air, due to its nonlinearity enabling to generate audio range acoustical waves or alternatively low frequency ultrasonic waves which can be used to characterize porous materials in the reflection configuration at normal incidence. Some appropriate theoretical calculations are introduced for three configurations of interest, i.e. a porous slab, a porous layer mounted onto a rigid plate, and a porous half space, in the case of the equivalent-fluid model. Comparisons between theoretical modeling and experimental data are provided and prospective industrial applications are discussed.     

一种基于非衍射波的高帧率超声成像发射系统的研究

基于非衍射波的高帧率(high frame rate,HFR)超声成像的前提是发射阵列波(array beam)声场,但由于阵列波的正弦函数特征使对超声发射信号加权处理变得比较困难,导致了发射系统复杂.针对这一问题,本文提出一种有效的解决方案.基于傅里叶变换理论,从方波的能量成分主要集中在基波这一特性出发,采用单值方波对超声传感器发射信号进行加权,用一个功率信号源和简单的电子开关网络实现了HFR系统中所需要的发射电路.实验表明,这种方案对HFR的成像质量几乎没有什么影响,但大大简化了HFR成像系统,这为HFR超声成像方法在现有B超声系统上的实现提供了一个可行的方法.     

基于 L-曲线参数优化的均匀声场重建算法

为了得到最小二乘法声场重建问题的稳定解,通常需要引入Tikhonov正则化方法。然而正则化程度取决于正则化参数的选择。针对这一问题,提出了一种基于L-曲线法参数选择的均匀声场重建算法。该算法根据重建误差与扬声器功率计算得到L-曲线,该曲线上曲率最大的点所对应的参数值作为Tikhonov正则化参数的选值。确定正则化参数后可进一步得到扬声器权系数以及重建均匀声场。针对不同正则化参数取值方法,对控制区域进行均匀声场重建以及重建性能仿真。仿真结果及实验表明,L-曲线法实现了重建误差与扬声器驱动信号功率之间的平衡。     

压电瞬态响应在换能器声功率测量中的应用

探索一种简便的聚焦超声功率测量方法,利用压电陶瓷片直接接收超声信号,通过机电类比得到压电瞬态响应由压电片在声波作用力下引起受迫振动产生的电压响应与固有振动产生的高频衰减响应叠加而成,分析输出压电信号与换能器声功率之间的换算关系。对输出压电信号进行二次包络提取,获得表征声功率变化的电压幅度曲线,分别找出不同换能器驱动电压下包络曲线的最大峰值电压,将其平方值与声功率计所测声功率进行线性拟合,并对理论关系式中的比例系数进行标定。实验结果所得线性拟合度较高,且标定后所得声功率与声功率计所测值相对误差低于8.7%,证明了通过压电瞬态响应测量换能器声功率具有可行性。      

Analysis and calibration of system errors in steerable parametric loudspeakers

By adjusting a set of delay amounts and amplitudes of the ultrasonic transducer (primary source) array in parametric loudspeakers, the directional sound beam can be steered within a range of predefined angles. This beamsteering characteristic of parametric loudspeakers has been proposed in theory and validated by measurements. In particular, the locations of the mainlobe and grating lobes can be predicted within a certain degree of accuracy in theory. However, errors incur in different stages of implementation. Thus, mismatches are observed between theoretical and measured beampatterns. In this paper, four types of system errors are analyzed for the primary-frequency waves and the difference-frequency waves based on the phased array theory and the product directivity principle, respectively. The degraded beampatterns which are caused by system errors are analyzed and calibrated by using a combined optimization approach of Monte Carlo method and nonlinear least squares method. Experimental results are presented to show the advantage of the proposed calibration method that leads to significant reduction of mismatch between theoretical and measured beampatterns at both the primary frequency and the difference frequency.     

经颅超声聚焦方法和声场特性研究

基于高分辨的CT数据建立了非均匀颅骨仿真模型,该模型引入了颅骨的声衰减系数,深入研究和分析了声波时间反转法和超声相控阵法在颅脑中的聚焦方法及效果。颅骨具有较强的声波衰减特性,使用时间反转聚焦时需要进行幅度补偿,对于0.7MHz的频率信号,幅度补偿后的时间反转聚焦声场主瓣宽度窄、旁瓣低,焦点处声场比无幅度补偿的时间反转法提高8.86dB,比超声相控阵聚焦法提高7.89dB,具有很好的空间聚焦精度和聚焦效率。研究了颅骨衰减系数、声场焦点位置、声波频率、换能器阵列位置和方位等参数对聚焦声场的影响,结果表明,幅度补偿时间反转法比相控阵法具有更低的旁瓣,且高频时的聚焦效果比相控阵好,相控阵聚焦对换能器阵列的位置和方位比较敏感,而时间反转经颅超声聚焦对声传播路径和入射角具有更高的鲁棒性。