Discrimination of sound source velocity in human listeners

The ability of six human subjects to discriminate the velocity of moving sound sources was examined using broadband stimuli presented in virtual auditory space. Subjects were presented with two successive stimuli moving in the frontal horizontal plane level with the ears, and were required to judge which moved the fastest. Discrimination thresholds were calculated for reference velocities of 15, 30, and 60 degrees/s under three stimulus conditions. In one condition, stimuli were centered on 0 degrees azimuth and their duration varied randomly to prevent subjects from using displacement as an indicator of velocity. Performance varied between subjects giving median thresholds of 5.5, 9.1, and 14.8 degrees/s for the three reference velocities, respectively. In a second condition, pairs of stimuli were presented for a constant duration and subjects would have been able to use displacement to assist their judgment as faster stimuli traveled further. It was found that thresholds decreased significantly for all velocities (3.8, 7.1, and 9.8 degrees/s), suggesting that the subjects were using the additional displacement cue. The third condition differed from the second in that the stimuli were "anchored" on the same starting location rather than centered on the midline, thus doubling the spatial offset between stimulus endpoints. Subjects showed the lowest thresholds in this condition (2.9, 4.0, and 7.0 degrees/s). The results suggested that the auditory system is sensitive to velocity per se, but velocity comparisons are greatly aided if displacement cues are present.     

采用听觉滤波器的宽带MUSIC声源定位方法

在分析了采用短时傅里叶变换的宽带MUSIC声源定位算法(SF-MUSIC)存在问题的基础上,提出了一种采用听觉滤波器的宽带MUSIC声源定位算法(AF-MUSIC)。该算法使用听觉滤波器组对传声器阵列接收到的信号进行不等带宽分解后,在各个频率通道上使用MUSIC算法进行声源定位,并结合子区间频数估计法得出最终定位结果。对算法进行的实验评估表明,在不同声源类型条件下,相比SF-MUSIC算法,AF-MUSIC算法的平均估计误差减少2.5479°,有效地提高了声源波达方向估计的精度。     

Auditory filter based broadband MUSIC algorithm for sound source localization

Based on the analysis of the shortcomings of broadband MUSIC algorithm with short-time Fourier transform （SF-MUSIC） for sound source localization, a broadband MUSIC algorithm with auditory filter （AF-MUSIC） was proposed. The proposed algorithm first em- ploys auditory filter bank to decompose the signals received on the microphone array, and then locates the sound source with MUSIC algorithm over every frequency channel. At last, by combining with the subinterval frequency estimation, the final localization result is gained. Evaluations on the proposed algorithm prove that comparing with the SF-MUSIC algorithm, the AF-MUSIC algorithm decreases the average error of the estimation results with 2.5479 de- gree in different source conditions. The accuracy of sound source DOA estimation is enhanced effectively.     

风场环境中声速修正的分布式声源定位算法

为减小声速误差对定位精度的影响,提出了一种基于声速修正的分布式声源定位方法。首先,将声速表示为未知声源位置的函数,逼近风场中的声速场分布,然后将其代入TDOA (Time Differences of Arrival)算法中,构建非线性超定方程组,最后采用粒子群优化算法求解声源位置。对不同风速、不同声源位置及不同测试区域进行仿真,结果表明:修正后的定位精度比修正前有明显提高,尤其对于大范围并且声源靠近测试区域边缘位置的定位系统,改善更加明显;4个节点的定位系统实验结果表明,修正后的定位误差可降至修正前的4l%,该方法能更好的应用于风场中的定位系统。     

Distributed sound source localization algorithm with sound velocity calibration in windy environments

A new sound source localization method with sound speed compensation is proposed to reduce the wind influence on the performance of conventional TDOA(Time Difference of Arrival) algorithms. First, the sound speed is described as a set of functions of the unknown source location, to approximate the acoustic velocity field distribution in the wind field. Then,they are introduced into the TDOA algorithm, to construct nonlinear equations. Finally, the particle swarm optimization algorithm is used to estimate the source location. The simulation results show that the proposed algorithm can significantly improve the localization accuracy for different wind velocities, source locations and test area sizes. The experimental results show that the proposed method can reduce localization errors to about 40% of the original error in a four nodes localization system.     

Interferometric method for estimating the sound source velocity in an oceanic waveguide

The noise immunity of an algorithm for estimating the radial component of sound source velocity, based on the data on frequency shifts of field interferencemaxima, has been investigated. A limiting estimate for the input signal-to-noise ratio, which determines the algorithm efficiency, is obtained. The results of computational and natural experiments are reported.     

抵消Wigner-Ville分布交叉项的窄带运动声源无源测速

利用声波的多普勒频移可以对窄带运动声源进行单传感器无源测速,其性能很大程度上取决于能否精确地估计出声波的瞬时频率.Wigner-Ville分布虽然时频分辨率高,但存在交叉项干扰,很少被直接用于瞬时频率估计。对此,提出了抵消Wigner-Ville分布交叉项的单传感器窄带声源无源测速方法。利用交叉项与声源速度的关系构造一个抵消项,引入到Wigner-Ville分布中,通过对声源速度估计值进行迭代更新,使抵消项与交叉项相位相反,从而约掉交叉项。经实测噪声数据验证,对一辆以6.07 m/s匀速运动的卡车(信噪比约为29 dB)测速误差为0.1 m/s,运行时间为4.6 s,对一架以28.90 m/s匀速运动的直升机(信噪比约为16 dB)测速误差为0.46 m/s,运行时间为1.2 s,均优于匹配Wigner变换和多普勒线性调频小波变换测速方法.     

Auditory profile analysis of irregular sound spectra

The discrimination of changes in the shapes of sound spectra is reported. The change was always an intensity increment to the 948-Hz component of a multitone complex. First, the ability of naive listeners to learn to discriminate a change in a "regular" background or reference spectrum (equal-level tones equally spaced in logarithmic frequency) was measured as a function of the number of trials. On the average, threshold improved about 10 dB over 3000 trials, with about 50% of the decrease in threshold occurring during the first 750 trials. In a subsequent series of experiments, the overall pattern of spectral shape of the background was varied randomly. Two kinds of perturbations in spectral shape were employed: Randomly choosing the frequencies of the reference spectra and randomly choosing the amplitudes of the components of the reference spectra. The experimental manipulations involved fixing the random spectra across a block of trials, varying the reference spectra from interval to interval of each trial, and providing extensive practice in discriminating specific randomly perturbed reference spectra. The results of the spectrum-learning and random perturbation experiments provide insight into the roles of critical band filtering, sensory variability, and short-term and long-term memory representations in auditory profile analysis. Further, the appropriateness of the generalization of a simple energy detection model is discussed.     

水中声速的测量

将空气中声速测量实验拓展至水中声速的测量,将NaCl溶于水中,制成不同浓度的盐水来模拟海水,对不同盐分的"海水"中的声速进行了测量.结果表明,随着海水盐度的增加,声速也增加.     

Directional perception of distributed sound sources

The perception of spatially distributed sound sources was investigated by conducting two listening experiments in anechoic conditions with 13 loudspeakers evenly distributed in the frontal horizontal plane emitting incoherent noise signals. In the first experiment, widely distributed sound sources with gaps in their distribution emitted pink noise. The results indicated that the exact loudspeaker distribution could not be perceived accurately and that the width of the distribution was perceived to be narrower than it was in reality. Up to three spatially distributed loudspeakers that were simultaneously emitting sound could be individually perceived. In addition, the number of loudspeakers that were indicated as emitting sound was smaller than the actual number. In the second experiment, a reference with 13 loudspeakers and test cases with fewer loudspeakers were presented and their perceived spatial difference was rated. The effect of the noise bandwidth was of particular interest. Noise with different bandwidths centered around 500 and 4000 Hz was used. The results indicated that when the number of loudspeakers was increased from four to seven, the perceived auditory event was very similar to that perceived with 13 loudspeakers at all bandwidths. The perceived differences were larger in wideband noise than in narrow-band noise.     

Interference immunity of an interferometric method of estimating the velocity of a sound source in shallow water

We describe an algorithm for estimating the radial component of the velocity of a sound source based on information about frequency shifts of the interference maxima of the field and consider the problem of its interference immunity. We obtain the limit estimate for the value of the input signal/noise ratio when the algorithm is working effectively. We present results of computational and field experiments using a single receiver and a horizontal array. We compare the experimental data with analytic interference immunity estimates.     

有限长线声源斜面声场分析

为了进一步研究有限长线声源的声场特性,完善有限长线声源声场理论,建立了有限长线声源斜面声场的模型,提出了其理论计算方法。基于仿真结果和数据分析,探讨并得到了有限长线声源斜面声场特性的三个参数与斜面声场特性之间的关系。通过调整斜面倾角α、交点的位置r0以及有限长线声源的长度L,可以有效改善斜面声场的分布。     

Underwater tunable organ-pipe sound source

A highly efficient frequency-controlled sound source based on a tunable high-Q underwater acoustic resonator is described. The required spectrum width was achieved by transmitting a linear frequency-modulated signal and simultaneously tuning the resonance frequency, keeping the sound source in resonance at the instantaneous frequency of the signal transmitted. Such sound sources have applications in ocean-acoustic tomography and deep-penetration seismic tomography. Mathematical analysis and numerical simulation show the Helmholtz resonator's ability for instant resonant frequency switching and quick adjustment of its resonant frequency to the instantaneous frequency signal. The concept of a quick frequency adjustment filter is considered. The discussion includes the simplest lumped resonant source as well as the complicated distributed system of a tunable organ pipe. A numerical model of the tunable organ pipe is shown to have a form similar to a transmission line segment. This provides a general form for the principal results, which can be applied to tunable resonators of a different physical nature. The numerical simulation shows that the "state-switched" concept also works in the high-Q tunable organ pipe, and the speed of frequency sweeping in a high-Q tunable organ pipe is analyzed. The simulation results were applied to a projector design for ocean-acoustic tomography.     

声速测量实验仪的改进

针对原有声速测量仪中存在的不足,利用单片机、步进电机等器件构建控制电路,通过识别信号强度强弱变化来代替原有实验仪中手动操作,提高了实验精度.     

Perception of the sound source position

This paper presents several experiments on sound source localization. They are based on monaural click presented at different interclick intervals (ICI), from 10 to 100 ms. Trains of clicks were presented to 10 healthy subjects. At short interclick intervals the clicks were perceived as a blur of clicks having a buzzy quality. Moreover, it was proven that the accurateness in the response improves with the increase of the length of ICI. The present results imply the usefulness of the interclick interval in estimating the perceptual accuracy. An important benefit of this task is that this enables a careful examination of the sound source perception threshold. This allows detecting, localizing and dividing with a high accuracy the sounds in the environment. The text was submitted by the authors in English.     

Auditory spectral integration in the perception of diphthongal vowels

This study considers an operation of an auditory spectral integration process which may be involved in perceiving dynamic time-varying changes in speech found in diphthongs and glide-type transitions. Does the auditory system need explicit vowel formants to track the dynamic changes over time? Listeners classified diphthongs on the basis of a moving center of gravity (COG) brought about by changing intensity ratio of static spectral components instead of changing an F2. Listeners were unable to detect COG movement only when the F2 change was small (160 Hz) or when the separation between the static components was large (4.95 bark).     

共振干涉法声速测定中声强减损的讨论

介绍子在共振干涉声速测定中,随接收探头和发射探头的距离增加,声强（能流密度）减小的规律和机制,指出了发射探头活塞式振动是导致能流密度减少的主要原因。