运动声源快速定位的声达时差法

针对声达时差法只能用于非运动声源定位的问题,本文提出一种运动声源快速定位方法。该方法以声达时差为基本定位原理,基于声源计算位置对多普勒效应进行解耦并进行声信号多普勒效应修正,根据三角定位方法构建声传播空间矩阵,以声源位置偏差度为目标基于单纯形优化搜索算法进行声源位置快速逼近,实现了对匀速直线运动的单声源的定位追踪,提高定位实时性。该方法将声达时差法拓展到运动声源的定位,同时解决了消除多普勒效应带来的计算过程复杂、运算量大的问题,仅用4个传声器就可实现运动声源的快速定位,突破了传统运动声源识别中对大传声器阵列的依赖。仿真实验和实车运动声源识别实验结果证明了该方法的有效性,本研究为短时发声运动声源的识别提供了一种简便、高效的方法。     

运动声源快速定位的声达时差法

针对声达时差法只能用于非运动声源定位的问题,本文提出一种运动声源快速定位方法。该方法以声达时差为基本定位原理,基于声源计算位置对多普勒效应进行解耦并进行声信号多普勒效应修正,根据三角定位方法构建声传播空间矩阵,以声源位置偏差度为目标基于单纯形优化搜索算法进行声源位置快速逼近,实现了对匀速直线运动的单声源的定位追踪,提高定位实时性。该方法将声达时差法拓展到运动声源的定位,同时解决了消除多普勒效应带来的计算过程复杂、运算量大的问题,仅用4个传声器就可实现运动声源的快速定位,突破了传统运动声源识别中对大传声器阵列的依赖。仿真实验和实车运动声源识别实验结果证明了该方法的有效性,本研究为短时发声运动声源的识别提供了一种简便、高效的方法。     

分布式浮标阵水下高速运动声源三维被动定位

给出一种用于垂直爬升弹道三维坐标精确测量的分布式浮标网络超长基线宽带被动定位技术。考虑四个呈正方形布局的浮标节点,利用水平方位角交汇的定位方法测量上浮弹道的水平x、y坐标,水平方位角估计由灵巧的声矢量传感器替代常规阵得到,并给出定向误差小于0.4°的湖试结果;利用直达声和海面反射声之间的时延差提取弹道的垂直坐标z,并给出了高速运动目标垂直爬升弹道的海试测量结果,弹道测量误差小于10 m。     

The influence of pinnae-based spectral cues on sound localization

The role of pinnae-based spectral cues was investigated by requiring listeners to locate sound, binaurally, in the horizontal plane with and without partial occlusion of their external ears. The main finding was that the high frequencies were necessary for optimal performance. When the stimulus contained the higher audio frequencies, e.g., broadband and 4.0-kHz high-pass noise, localization accuracy was significantly superior to that recorded for stimuli consisting only of the lower frequencies (4.0- and 1.0-kHz low-pass noise). This finding was attributed to the influence of the spectral cues furnished by the pinnae, for when the stimulus composition included high frequencies, pinnae occlusion resulted in a marked decline in localization accuracy. Numerous front-rear reversals occurred. Moreover, the ability to distinguish among sounds originating within the same quadrant also suffered. Performance proficiency for the low-pass stimuli was not further degraded under conditions of pinnae occlusion. In locating the 4.0-kHz high-pass noise when both, neither, or only one ear was occluded, the data demonstrated unequivocally that the pinna-based cues of the "near" ear contributed powerfully toward localization accuracy.     

基于近场声全息理论的运动声源动态识别方法

运动声源的准确识别是对其进行有效控制的重要前提,本文在近场声全息理论基础上,利用时域多普勒效应消除原理、球面波传播原理,建立了用于运动噪声源识别的动态识别方法。模拟计算和实车试验证明该识别方法能对高速运动噪声源进行准确识别。     

水下运动声源的方位方差检测方法

为了提高对水下运动目标的检测能力,利用水下目标辐射信号与海洋环境噪声在空间分布特性上的差异,提出了一种方位方差检测方法。该方法采用时延估计法估计目标方位,对方位求方差以获得检验统计量。采用奈曼-皮尔逊准则,推导了该方法的检测概率关系式,得出了检测性能与信噪比关系曲线,并与宽带能量检测方法进行了比较,表明该方法的性能远优于宽带能量检测方法。理论分析和海上试验结果都验证了该方法的有效性。     

The effect of multimicrophone noise reduction systems on sound source localization by users of binaural hearing aids

This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.     

The influence of duration and level on human sound localization

The localization of sounds in the vertical plane (elevation) deteriorates for short-duration wideband sounds at moderate to high intensities. The effect is described by a systematic decrease of the elevation gain (slope of stimulus-response relation) at short sound durations. Two hypotheses have been proposed to explain this finding. Either the sound localization system integrates over a time window that is too short to accurately extract the spectral localization cues (neural integration hypothesis), or the effect results from cochlear saturation at high intensities (adaptation hypothesis). While the neural integration model predicts that elevation gain is independent of sound level, the adaptation hypothesis holds that low elevation gains for short-duration sounds are only obtained at high intensities. Here, these predictions are tested over a larger range of stimulus parameters than has been done so far. Subjects responded with rapid head movements to noise bursts in the two-dimensional frontal space. Stimulus durations ranged from 3 to 100 ms; sound levels from 26 to 73 dB SPL. Results show that the elevation gain decreases for short noise bursts at all sound levels, a finding that supports the integration model. On the other hand, the short-duration gain also decreases at high sound levels, which is in line with the adaptation hypothesis. The finding that elevation gain was a nonmonotonic function of sound level for all sound durations, however, is predicted by neither model. It is concluded that both mechanisms underlie the elevation gain effect and a conceptual model is proposed to reconcile these findings.     

The influence of signal parameters on the sound source localization ability of a harbor porpoise (Phocoena phocoena)

It is unclear how well harbor porpoises can locate sound sources, and thus can locate acoustic alarms on gillnets. Therefore the ability of a porpoise to determine the location of a sound source was determined. The animal was trained to indicate the active one of 16 transducers in a 16-m-diam circle around a central listening station. The duration and received level of the narrowband frequency-modulated signals (center frequencies 16, 64 and 100 kHz) were varied. The animal's localization performance increased when the signal duration increased from 600 to 1000 ms. The lower the received sound pressure level (SPL) of the signal, the harder the animal found it to localize the sound source. When pulse duration was long enough (approximately 1 s) and the received SPLs of the sounds were high (34-50 dB above basic hearing thresholds or 3-15 dB above the theoretical masked detection threshold in the ambient noise condition of the present study), the animal could locate sounds of the three frequencies almost equally well. The porpoise was able to locate sound sources up to 124 degrees to its left or right more easily than sounds from behind it.     

Material sound source localization through headphones

In the present paper a study of sound localization is carried out, considering two different sounds emitted from different hit materials (wood and bongo) as well as a Delta sound. The motivation of this research is to study how humans localize sounds coming from different materials, with the purpose of a future implementation of the acoustic sounds with better localization features in navigation aid systems or training audio-games suited for blind people. Wood and bongo sounds are recorded after hitting two objects made of these materials. Afterwards, they are analysed and processed. On the other hand, the Delta sound (click) is generated by using the Adobe Audition software, considering a frequency of 44.1 kHz. All sounds are analysed and convolved with previously measured non-individual Head-Related Transfer Functions both for an anechoic environment and for an environment with reverberation. The First Choice method is used in this experiment. Subjects are asked to localize the source position of the sound listened through the headphones, by using a graphic user interface. The analyses of the recorded data reveal that no significant differences are obtained either when considering the nature of the sounds (wood, bongo, Delta) or their environmental context (with or without reverberation). The localization accuracies for the anechoic sounds are: wood 90.19%, bongo 92.96% and Delta sound 89.59%, whereas for the sounds with reverberation the results are: wood 90.59%, bongo 92.63% and Delta sound 90.91%. According to these data, we can conclude that even when considering the reverberation effect, the localization accuracy does not significantly increase.     

循环平稳声场的声源定位研究

旋转机械产生的辐射声场具有循环平稳特性,传统的平面近场声全息技术无法准确反映其调制特性,往往在边频带上出现虚假的能量的分布.采用循环谱密度取代功率谱密度作为重建物理量,则可准确提取循环平稳声场的调制和载波信息.考虑到循环谱密度的计算量以及特征提取的准确性,提出了循环谱密度组合切片分析法,并分析了加性白噪声对重建的影响.仿真分析及实验结果表明,此方法有较强的噪声抑制能力,全息重建的结果可准确反映声源的位置.     

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。     

A study on the random array generation method for moving sound source identification

Based on the problem that the generating method of random array structure is inefficient,a method is proposed to generate the random target arrays by using coaxial circular array in the polar coordinates in the premise that the array angular resolution of source identification is guaranteed.According to the principle of moving sound source identification,this work deduces the basic non-equidistance coaxial circular rings'radius,and generates target random arrays which were suitable for moving sound source identification through array partitioning,condition filtering in the polar coordinates and simulation evaluation.Finally,numerical simulation and moving car sound source identification test have been done.The analytical results show that using this method to generate random array is effective.Compared with the traditional regular arrays,the target random array has more accurate moving sound source identification performance.     

高精度球面阵聚焦声源定位方法研究

提出了一种高精度高空间分辨率球面阵聚焦声源定位方法——虚拟源法。该方法通过球面阵波束扫描获得实际声源的空间聚焦谱,并假定各扫描点为虚拟声源,将实际声源聚焦谱看作是全体虚拟源共同作用的结果,由此得到各虚拟源对声场的贡献量,从而可实现声源精确定位。仿真研究分析了频率,阵列孔径,声场模态阶数,信噪比等参数对声源定位性能的影响,并与常规算法进行对比。结果显示,该方法不受频率和阵列孔径的限制,避免了空间“混淆”,能够进行高精度高分辨率声源定位,并具有良好的背景噪声抑制能力。     

频率对环绕声声像定位的影响

本文考虑双耳相位差的高级近似，导出了中频情况下适用的具有更普遍意义的平面环绕声声像定位公式。在低频时该式将化为通常的环绕声声像定位公式，而随着声音频率的增加，声像位置将与频率有关。将新的公式用到方型排列和棱型排列的４－４－４环绕声系统，得到了同实验相一致的结果。文中着重指出，声像随频率而变化是导致环绕声重发中侧向声像不稳定的重要在而为今后改进环绕声系统提供了理论基础。     

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。     

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。     

Non-line-of-sight sound source localization using matched-field processing

Acoustic diffraction allows sound to travel around opaque objects and therefore may allow beyond-line-of-sight sensing of remote sound sources. This paper reports simulated and experimental results for localizing sound sources based on fully shadowed microphone array measurements. The generic geometry includes a point source, a solid 90° wedge, and a receiving array that lies entirely in the shadow defined by the source location and the wedge. Source localization performance is assessed via matched-field (MF) ambiguity surfaces as a function of receiving array configuration, and received signal-to-noise ratio for the Bartlett and minimum variance distortionless (MVD) MF processors. Here, the sound propagation model is developed from a Green's function integral treatment. A simple 16 element line array of microphones is tested in three mutually orthogonal orientations. The experiments were conducted using an approximate 50-to-1-scaled tabletop model of a blind city-street intersection and produced ambiguity surfaces from source frequencies between 17.5 and 19 kHz that were incoherently summed. The experimental results suggest that a sound source may be localized by the MVD processor when using fully shadowed arrays that have significant aperture parallel to the edge of the wedge. However, this performance is reduced significantly for signal-to-noise ratios below 40 dB.     

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。     

Fly-ear inspired micro-sensor for sound source localization in two dimensions

Inspired by the hearing organ of the fly Ormia ochracea, a miniature sound localization sensor is developed, which can be used to pinpoint a sound source in two dimensions described by the azimuth and elevation angles. The sensor device employs an equilateral triangle configuration consisting of three mechanically coupled circular membranes whose oscillations are detected by a fiber-optic system. The experimental results indicate that significant amplification of the directional cues and directional sensitivity can be achieved with the fly-ear inspired sensor design. This work can provide a basis for the development of miniature sound localization sensors in two dimensions.