Noise source localization in shallow water

The possibilities of the method for detecting a noise source, estimating its radial velocity and its distance from the receiver, based on the double Fourier transform of the interference pattern formed during motion, have been considered for a single-receiver configuration. It is shown that, as compared to a signal with a uniform spectrum, the limiting values of the input signal-to-noise ratio, at which the noise source can be localized, increase by about half. The results of a computational experiment are presented. A comparative analysis of the noise immunity of the algorithm applied to signals with uniform and noise spectra is performed.     

Tracking-positioning of sound speed profiles and moving acoustic source in shallow water

An ensemble Kalman filter(EnKF) approach is proposed to perform sequential tracking of water column sound speed profile(SSP) using a moving acoustic source. First,the SSPs are discretized in depth and range, and are expressed by the empirical orthogonal functions(EOFs). Second, the acoustic source state information and the first three orders of EOF coefficients are expressed as the state variable, and the acoustic field information received by the vertical line array are the measured values. Successively, the state variables and measured values are used to establish the state-measure model. Last, the EnKF is utilized to track the state variables. The simulation results show that the root mean square error of SSP and the absolute error of source are all small, and thus the acoustic source tracking-positioning has high accuracy. Moreover, increasing the number of sample collection, the signal-to-noise ratio and the number of receiving elements can improve the tracking-positioning results. The method is verified using the experimental data of the East China Sea.     

浅海声速剖面与移动声源的跟踪定位

在水平非均匀分布的浅海环境中,针对移动声源跟踪时,声速剖面的变化会对声场产生影响,提出了一种利用集合卡尔曼滤波算法的声速剖面跟踪反演和移动声源跟踪定位的方法。首先,将声速剖面进行距离和深度的参数化表示,从而将对声速剖面的跟踪转化为对声速剖面前3阶经验正交函数系数的跟踪;其次,通过将声源状态信息和声速剖面信息表示为状态变量,而将垂直线列阵接收到的声场信息作为测量值建立状态-测量模型,然后利用集合卡尔曼滤波方法对模型状态变量进行跟踪。仿真结果得出:声速剖面跟踪反演的均方根误差和移动声源跟踪定位的绝对误差都非常小,对声源的跟踪定位精度很高。并且通过增加集合样本数、增加接收信号信噪比以及增加接收阵元数目都可以提高跟踪定位结果精度。最后,利用东海实验数据对本方法进行了验证。     

浅海单矢量传感器宽带声源三维定位仿真

针对浅海水平不变波导中利用单矢量传感器的低频宽带声源三维被动定位问题,首先利用平均声强器估计声源方位;其次,通过分离后的简正模声压和水平振速分量联合处理获得运动声源相对速度,进一步基于垂直复声强中简正模相干项特征频率不变性以及长时间窗口中多快拍信号的统一处理,建立以warping变换归一化频谱强度和作为代价函数的距离搜索处理器,估计该段信号的初始距离,进而获得各时刻声源距离,所提出的方法避免了对拷贝声场和引导声源的依赖。信噪比大于10 dB时,相对误差在11.33%之内;最后,利用多阶简正模相干项与非相干项能量的模基处理方法,当声场中存在三阶以上简正模时,对声源深度进行匹配估计。仿真分析表明,单个矢量传感器能够完成声源的方位、距离及深度的三维定位。     

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.     

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.     

浅水海豚三维空间定位与发声源级测量

详细阐述了单水听器和水听器阵列进行海豚定距定位的几何模型,以及利用不同接收通道间的时延差计算空间坐标的方法,结合实验数据计算了海豚在水中的三维空间坐标和回声定位信号的声源级,并开展了海豚运动轨迹追踪分析。结果表明,相对于单水听器,水听器阵列对测量环境和海豚行为的限制较少,但对采集设备的同步性和硬件连接有较高要求。误差分析表明水听器阵列在大于3 m的距离,定位误差可降低到5%以下。研究结果为精确计算海豚发声源级及海豚的声呐性能及行为研究提供了技术支撑。     

浅水海豚三维空间定位与发声源级测量

详细阐述了单水听器和水听器阵列进行海豚定距定位的几何模型,以及利用不同接收通道间的时延差计算空间坐标的方法,结合实验数据计算了海豚在水中的三维空间坐标和回声定位信号的声源级,并开展了海豚运动轨迹追踪分析。结果表明,相对于单水听器,水听器阵列对测量环境和海豚行为的限制较少,但对采集设备的同步性和硬件连接有较高要求。误差分析表明水听器阵列在大于3 m的距离,定位误差可降低到5%以下。研究结果为精确计算海豚发声源级及海豚的声呐性能及行为研究提供了技术支撑。     

Focusing of low-frequency sound fields in shallow water

A numerical experiment is carried out to study the focusing of a low-frequency (100–300 Hz) sound field in a shallow-water acoustic waveguide typical of an oceanic shelf. Focusing with the use of time reversal of broadband acoustic signals, which is called time reversal mirror (TRM) of waves, is considered along with focusing by phase conjugation (PC) of a monochromatic sound field. It is demonstrated that, in the case of focusing by the TRM method in the waveguide of interest, it is sufficient to have a single source-receiving element. The use of a vertical array improves the quality of focusing. The quality achieved in the latter case proves to be approximately the same as that achieved in the case of focusing by phase conjugation of a monochromatic field at a frequency identical to the carrier frequency of the broadband signals. It is also shown that, in a range-independent waveguide, intense surface waves considerably reduce the quality of focusing. This effect is most pronounced in the case of using phase conjugation.     

Horizontal refraction of sound in a shallow water and its experimental observations

Basic results of theoretical and experimental studies of acoustic effects caused by horizontal refraction in a shallow water are reviewed with special emphasis on the effect produced by intense internal waves on the propagation of broadband signals. Theoretical calculations and experimental data testifying to the manifestation of three-dimensional acoustic effects in the SWARM’95 experiment are presented.     

The characteristic of sound propagation in deep water and underwater source localization in the direct zone

The multiple-path sound propagation in deep water is conducive to source localization of an underwater target.The transmission losses(TLs) and broadband pulse multiple-path propagation characteristics from a deep receiver is analyzed by using the experimental data from deep water area in the South China Sea(SCS).The results indicate that the width of the direct zone near the bottom of 4300 m water depth is about 30 km.The TLs in the direct zone near the bottom are much less than those in the shadow zone.It is meaningful for underwater sound source detection.Moreover,the time delay between the direct path and the bottomsurface-reflected path for a receiver near the bottom decreases monotonically with the source range.According to the linear relationship between the time delay of multipath and source range,a source localization method is presented to estimate the range of underwater target.The experimental results show that the estimated ranges are consistent with the global position system(GPS) measurements,and the mean square error of the estimation results is less than 0.28 km.     

Broadband sound propagation in shallow water and geoacoustic inversion

Part of an experiment to test a measurement package in a shallow water region in the Gulf of Mexico was designed to gather broadband acoustic data suitable for inversion to estimate seabed geoacoustic parameters. Continuous wave tow acoustic signals at multiple frequencies and broadband impulsive source signals were recorded on a horizontal line array in a high-noise environment. Simulated annealing with a normal mode forward propagation model is utilized to invert for a geoacoustic representation of the seabed. Several inversions are made from different data samples of two light bulb implosions, the measured sound speed profiles at the HLA and at the positions of the light bulb deployments, and for two different cost functions. The different cost functions, measured sound speed profiles, and measured time series result in different inverted geoacoustic profiles from which transmission loss is generated for comparison with measurements. On the basis of physical consistency and from the comparison of the transmission loss and time series, a best estimate geoacoustic profile is selected and compared to those obtained from previously reported inversions. Uncertainties in the sound speed profile are shown to affect the uncertainties of the estimated seabed parameters.     

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.     

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 velocity of a noise sound source and the distance to it in shallow water using a vector-scalar receiver

An experiment on estimating the velocity of a noise source and the distance to it using a single vector-scalar receiver has been performed on shallow-water Pacific shelf. Expressions for the components of the vector-scalar receiver field are derived. The source parameters are reconstructed using the interferometric method. The noise immunity of the method is analyzed for different acoustic field components and their combinations. The sensitivity of the method with respect to changes in the bottom parameters is considered.     

Surface reverberation at sound field focusing in shallow water

A numerical experiment is carried out to study the long-range surface reverberation in the presence of intense surface waves for the case of using vertical transmitting arrays providing sound field focusing at different depths. To focus the field, a phase conjugation of acoustic waves from a probe source positioned at the focusing point is used. It is demonstrated that surface waves considerably affect the focusing quality at a distance of several tens of kilometers from the transmitting array. This prevents the efficient suppression of long-range reverberation by increasing the focusing depth.     

Three-dimensional sound source localization using distributed microphone arrays

To improve the performance of sound source localization based on distributed microphone arrays in noisy and reverberant environments,a sound source localization method was proposed.This method exploited the inherent spatial sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing(CS) theory.In this method two-step discrete cosine transform(DCT)-based feature extraction was utilized to cover both short-time and long-time properties of the signal and reduce the dimensions of the sparse model.Moreover,an online dictionary learning(DL) method was used to dynamically adjust the dictionary for matching the changes of audio signals,and then the sparse solution could better represent location estimations.In addition,we proposed an improved approximate l_0norm minimization algorithm to enhance reconstruction performance for sparse signals in low signal-noise ratio(SNR).The effectiveness of the proposed scheme is demonstrated by simulation results where the locations of multiple sources can be obtained in the noisy and reverberant conditions.     

一种利用分布式传声器阵列的声源三维定位方法

为了提高噪声和混响条件下分布式传声器阵列进行声源定位的性能,提出一种利用空间稀疏性和压缩感知原理的声源三维定位方法。该方法首先通过两次离散余弦变换方式提取出声音信号特征,并用该特征来构建稀疏定位模型,以便能够综合利用语音信号的短时和长时特性,同时降低模型维数;然后利用在线字典学习技术动态调整字典,克服稀疏模型与实际信号之间的失配问题,增强稀疏定位模型的鲁棒性;进而提出一种改进的平滑l₀范数稀疏重构算法来进行声源位置解算,以提高低信噪比条件下的重构精度。仿真结果表明该方法不仅可以实现多目标定位,而且具有较强的抗噪声和抗混响能力.