水声信号稀疏重构高分辨角度-多普勒成像

针对傅氏空时二维谱估计分辨率低以及声呐空时采样数据样本数不足给角度-多普勒成像带来困难的问题,提出一种水声信号稀疏重构的高分辨角度-多普勒成像方法和抗混响空时滤波器的稀疏重构方法。该方法在声呐阵列单测量向量的极少观测样本条件下,建立阵列信号的空时稀疏表示模型,应用稀疏表示的匹配追踪算法和基追踪算法重构回波与混响的高分辨角度-多普勒像。并根据运动声呐回波与混响的空时分布规律及声呐待检测距离单元位置的先验信息,沿着混响空时分布脊线设计混响稀疏表示的专用空时导向向量字典,通过重构抗混响空时滤波器来抑制角度-多普勒平面的混响干扰。对运动声呐前视和侧视阵列的计算机仿真结果表明,在混响背景中,该方法采用声呐阵列单测量向量重构了低速运动目标多亮点回波的高分辨角度-多普勒像,频率分辨率突破傅里叶分辨率,角度分辨率突破阵列瑞利限,分辨率明显优于傅氏空时谱估计。     

High-resolution angle-Doppler imaging by sparse recovery of underwater acoustic signals

The low resolution of Fourier two-dimensional spatial temporal spectrum estimation and the insufficient sample size of sonar space time sampling data often caused difficulties in high-resolution space time spectrum estimation. Aiming to solve this problem, we proposed a high-resolution angle-Doppler imaging method and designed an anti-reverberation space time filter based on the sparse recovery of underwater acoustic signals. The proposed imaging method established the spatial temporal sparse re...     

一种单频水声信号多径时延估计算法

提出一种针对单频水声信号的多径时延估计算法,通过采用频域加权并结合进化规划算法解决代价函数强烈振荡下的高效全局优化。算法引入多径径数建立由多径时延、幅度系数及径数组成的水声信道多径参数模型,在采用频域加权预白化进行非线性最小二乘代价函数解耦处理的基础上,结合进化规划算法进行非线性参数解空间的高效全局寻优,从而实现多径时延的高分辨率估计。利用仿真数据和海上实验数据进行的实验表明,本算法具有较高的多径时延估计精度并可同时实现多径径数的估计。     

Multipath time delay estimation of underwater acoustic sinusoidal signals

To overcome the performance limitation of multipath time-delay estimation posed by underwater acoustic sinusoidal signals, an approach incorporating the frequency-domain weighting of the highly oscillatory Nonlinear Least Squares （NLS） cost function with the evolu- tionary optimization was proposed to facilitate the accurate estimation of the multipath timedelay of sinusoidal signals. In the described method, the number of the effective multipath signals, which is included into the parameter model as well as the multipath time-delay and amplitude factor, can be estimated simultaneously thus avoiding the requirement of additional computation. The experimental results performed with numerical simulation and sea-trial data are provided, demonstrating the effectiveness and precision enhancement of the proposed algorithm.     

Experimental demonstration of underwater acoustic communication using bionic signals

This paper applies dolphin whistles to covert underwater acoustic (UWA) communication and proposes a UWA communication scheme based on M-ary bionic signal coding. At the transmitter end, the scheme maps multiple information bits into a dolphin whistle through a signal selector. At the receiver end, passive time reversal mirror (PTRM) is used for channel equalization and source information is restored according to the decision of which whistle is transmitted. The scheme has high spread spectrum gain. The anti multi-path performance is greatly improved when using PTRM. Different from traditional covert UWA communication methods, this mimicked signal is unlikely to alert an adversary even in high SNRs because of its real existence in marine environment. A tank experiment is conducted for the scheme, at communication rate of 50 bit/s with SNR −5 dB user information is recovered at a very low bit error rate. The results of tank experiment demonstrate the feasibility of this covert UWA communication scheme.     

Correlation of signals of thermal acoustic radiation

The spatial correlation function is measured for the pressure of thermal acoustic radiation from a source (a narrow plasticine plate) whose temperature is made both higher and lower than the temperature of the receiver. The spatial correlation function of the pressure of thermal acoustic radiation is found to be oscillatory in character. The oscillation amplitude is determined not by the absolute temperature of the source but by the temperature difference between the source and the receiver. The correlation function changes its sign when a source heated with respect to the receiver is replaced by a cooled one.     

Correlation of acoustic signals at summary delays

Correlation of acoustic signals received by spatially separated transducers from a single spatial region is analyzed. By additional experiments, it is shown that the origin of signal correlation at summary delays lies in the interference effects that occur in thermal acoustic noise in the space between the acoustic transducers.     

用独立成份分析算法实现水声信号盲分离

独立成份分析算法是在研究信号盲分离过程中出现的一种新方法,本文试图将几种独立成份分析算法用于分离水声信号。分析和比较了5种算法的性能,并用仿真信号对算法进行了仿真,阐述了独立成份分析算法分离水声信号的不足。针对含噪声模型,提出了一种基于独立成份分析算法成功分离水声信号的方法。     

水声信号稀疏重构的高阶累积量波达方向估计

高阶累积量具有高斯噪声抑制和阵元扩展特性,将高阶累积量引入水声信号的方位估计中,提出了离格稀疏贝叶斯学习重构的高阶累积量测向算法。该方法利用高阶累积量对高斯噪声的自然盲性,计算阵列信号四阶累积量来滤除高斯噪声,使阵元在原来的结构上扩展了一倍;并构造出选择矩阵剔除了四阶累积量中的冗余项,能再一次的扩展阵元,得到的新观测模型具有更好的统计性能;最后利用空域稀疏性,推导出四阶累积量下的离格稀疏表示模型,采用贝叶斯学习解算出源信号的最大后验概率,实现了目标方位估计。数值仿真和海试实验数据表明,该方法在相邻声源方位间隔为4°的情况下分辨概率可达到95%以上,在信噪比大于-5 dB时目标方位估计的均方根误差在1°以内,可显著抑制背景噪声干扰,在多声源密集分布条件下也能准确、稳健的对水声目标方位进行估计。     

Direction of arrival estimation based on high-order cumulant by sparse reconstruction of underwater acoustic signals

The high-order cumulant with the characteristics of Gaussian noise suppression and array expansion, can be introducd into the direction of arrival(DOA) estimation of the underwater acoustic signals. A DOA estimation algorithm based on high-order cumulant by off-grid sparse Bayesian learning is proposed. Considering the natural blindness of the highorder cumulant to Gaussian noise, the fourth-order cumulant of the array signal is calculated to filter out the Gaussian noise, with the doubled array...     

定向传输水声通信网络介质访问控制协议

针对水声通信网络吞吐量低、通信隐蔽性差、隐藏终端等问题,提出一种定向传输水声通信网络介质访问控制(Medium Access Control,MAC)协议。协议中各节点采用定向模式传输数据,不需要邻节点位置等先验信息。通过发送节点顺序连续发送传输请求(Request To Send,RTS)信号、维护邻节点相对位置信息表及定向虚拟载波监测等技术实现网络高效无碰撞运行。本协议可有效解决水声通信网络中存在的由非对称增益导致的隐藏终端问题,增加网络覆盖范围,提高通信隐蔽性。仿真结果表明,所提MAC协议能够显著提高水声通信网络吞吐量性能。     

Efficient processing of acoustic signals for high-rate information transmission over sparse underwater channels

For underwater acoustic channels where multipath spread is measured in tens of symbol intervals at high transmission rates, multichannel equalization required for bandwidth-efficient communications may become prohibitively complex for real-time implementation. To reduce computational complexity of signal processing and improve performance of data detection, receiver structures that are matched to the physical channel characteristics are investigated. A decision-feedback equalizer is designed which relies on an adaptive channel estimator to compute its parameters. The channel estimate is reduced in size by selecting only the significant components, whose delay span is often much shorter than the multipath spread of the channel. Optimal coefficient selection (sparsing) is performed by truncation in magnitude. This estimate is used to cancel the post-cursor ISI prior to linear equalization. Spatial diversity gain is achieved by a reduced-complexity pre-combining method which eliminates the need for a separate channel estimator/equalizer for each array element. The advantages of this approach are reduction in the number of receiver parameters, optimal implementation of sparse feedback, and efficient parallel implementation of adaptive algorithms for the pre-combiner, the fractionally-spaced channel estimators and the short feedforward equalizer filters. Receiver algorithm is applied to real data transmitted at 10 kbps over 3 km in shallow water, showing excellent results.     

Performance analysis of direct-sequence spread-spectrum underwater acoustic communications with low signal-to-noise-ratio input signals

Direct-sequence spread-spectrum signals collected from the TREX04 experiment are analyzed to determine the bit-error-rate (BER) as a function of the input signal-to-noise ratio (SNR) for a single receiver. A total of 1160 packets of data are generated by adding ambient noise data collected at sea to the signal data (in postprocessing) to create signals with different input-SNR, some as low as -15 dB. Two methods are analyzed in detail, both using a time-updated channel impulse-response estimate as a (matched) filter to mitigate the multipath-induced interferences. The first method requires an independent estimate of the time-varying channel impulse-response function; the second method uses the channel impulse-response estimated from the previous symbol as the matched filter. The first method yields an average BER <10(-2) for input-SNR as low as -12 dB and the second method yields a similar performance for input-SNR as low as -8 dB. The measured BERs are modeled using the measured signal amplitude fluctuation statistics and processing gain obtained by de-spreading the received signal with the transmitted code sequence. Performance losses caused by imprecise symbol synchronization at low input-SNR, uncertainty in channel estimation, and signal fading are quantitatively modeled and compared with data.     

Visualization of acoustic wavefronts by means of acoustic stroboscopy

The amplitude and phase distributions of a progressive acoustic wave are visualized simultaneously by means of acoustic stroboscopy. Two methods are described: (i) modulation of a laser beam by a progressive acoustic wave; (ii) modulation by a stationary wave. Spatial filtering of diffracted orders enables the progressive wavefronts to be seen as if they were stationary.     

Excitation of low-mode-number acoustic signals by a vertical array in shallow sea in the presence of underwater flows

We study the features of formation of low-mode-number acoustic signals in shallow sea by a vertical array matched to the hydroacoustic waveguide structure in the presence of drowned flows that cause the array bending and deflection from the vertical. For other given parameters of the waveguide and the flows, the efficiency of excitation of the low-mode-number signals is shown to be determined by the array position and length. It is demonstrated that the flow-induced array bending and deflection from the vertical introduces inhomogeneity in the structure of the excited low-mode-number fields. In particular, spatial modulation of the radiated signals can appear in the horizontal plane. Possibilities of reduction of the field inhomogeneity are discussed. __________ Translated from Izvestiya Vysshikh Uchebnykh Zavedenii, Radiofizika, Vol. 49, No. 11, pp. 925–941, November 2006.     

Correlation characteristics of pseudonoise signals in the ocean at a highly directional reception

Data of experiments with single-path and multipath reception of broadband continuous pseudonoise acoustic signals (0.8–1.3 kHz) in a deep ocean are discussed. The experiments were carried out on tracks from ～ 60 to ～ 420 km in length, which corresponds to the first seven convergence zones. The data on the cross-correlation coefficients and arrival time differences are presented (for the first, second, and seventh zones) for signals received by one highly directional (～2°) vertical array or by two such arrays simultaneously at depths of 200 and 450 m. The correlation characteristics of signals propagating over purely water paths, i.e., without reflections from the waveguide boundaries, and arriving under different angles in the vertical plane are compared. It is found that the origin of the decorrelation of signals received from various directions or at different depths (all other factors being the same) is the incomplete resolution of the received multipath signals in the arrival angle in the vertical plane.     

低频水下声信号的激光探测

根据表面波声光效应的原理,提出了一种低频水下声信号的激光探测技术,并建立了实验装置。在几十赫兹的低频段,对水下声源所产生的表面声波进行了探测。实验过程中,利用MATLAB软件对拍摄的衍射图样进行扫描分析,得到了衍射图样中条纹的像素差。根据波长与条纹间距的解析关系式,得到了低频液体表面声波波长,其大小在毫米量级。利用计算机编程,根据最小二乘法拟合色散关系的回归曲线,测量结果与理论色散关系吻合。该方法具有实时、非接触的特点。     

基于水声信道传播时延排序的分层空时信号检测

基于分层空时编码的多输入多输出技术是一种极具潜力的高速水声通信技术, 但要实现这种潜力需要复杂的空时信号处理方法, 以抵消来自水声信道的多径干扰和异步到达干扰, 以及叠加在接收端的各层信号之间的干扰. 对低复杂度的空时信号处理方案进行了研究, 提出了一种基于子信道传播时延排序的有序连续干扰抵消信号检测算法, 利用子信道间的传播时延差, 实现可使差错概率最小的最佳检测排序; 给出了利用信道估计, 以极低的计算量确定排序的方法, 从而可以大幅降低信号检测的计算复杂度. 采用低复杂度的单载波频域均衡来抵消水声信道中的码间干扰和异步到达干扰. 仿真结果表明, 基于时延排序的信号处理算法可以获得检测性能的改善, 而且性能增益在高数据率时更加显著. 研究结果表明, 采用有效的信号处理方法可使水声信道中造成信号检测干扰的传播时延成为改善系统性能的有利因素.     

Comparisons of spectral characteristics of wind noise between omnidirectional and directional microphones

Wind noise reduction is a topic of ongoing research and development for hearing aids and cochlear implants. The purposes of this study were to examine spectral characteristics of wind noise generated by directional (DIR) and omnidirectional (OMNI) microphones on different styles of hearing aids and to derive wind noise reduction strategies. Three digital hearing aids (BTE, ITE, and ITC) were fitted to Knowles Electronic Manikin for Acoustic Research. They were programmed to have linear amplification and matching frequency responses between the DIR and OMNI modes. Flow noise recordings were made from 0° to 360° azimuths at flow velocities of 4.5, 9.0, and 13.5 m/s in a quiet wind tunnel. Noise levels were analyzed in one-third octave bands from 100 to 8000 Hz. Comparison of wind noise revealed that DIR generally produced higher noise levels than OMNI for all hearing aids, but it could result in lower levels than OMNI at some frequencies and head angles. Wind noise reduction algorithms can be designed to detect noise levels of DIR and OMNI outputs in each frequency channel, remove the constraint to switch to OMNI in low-frequency channel(s) only, and adopt the microphone mode with lower noise levels to take advantage of the microphone differences.     

逆向傅里叶衍射定理快速预报水下目标三维声散射指向分布

提出逆向运用傅里叶衍射定理预报水下弱散射目标三维声散射指向分布的快速计算方法。依据目标形状、周围介质的密度和声速构建三维声场图像模型,建立散射远场积分结果与图像频域幅值的关系式,提取频域中半径为水中波数k;的球型表面上的幅值,获得精细化的宽带、全方位散射声压指向特性。数值计算表明:将傅里叶衍射定理逆向运用于解决声学正问题,适用于分层的、不均匀的、非规则及多体弱散射目标散射声场的求解。通过插值提取频域样本获得远场声压的方法,避免了有限元法(3D-FEM)所必须的大规模的网格划分和迭代运算,可以有效地减少计算成本并拓展散射频率响应的带宽。在水池中完成两种具有不同声学参数和形状目标指向性测试实验,得到散射声压指向性幅度函数与理论预报相一致。