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1.
Monitoring the thickness changes of channel siltation is paramount in safeguarding navigation and guiding dredging.This paper presents a novel method for realizing the field monitoring of channel siltation in real time.The method is based on the bistatic scattering theory and concerned more with the receiving and processing of multipath signal at high-frequency and small grazing angle.By use of the multipath propagation structure of underwater acoustic channel,the method obtains the silt thickness by calculating the relative time delay of acoustic signals between the direct and the shortest bottom reflected paths.Bistatic transducer pairs are employed to transmit and receive the acoustic signals,and the GPS time synchronization technology is introduced to synchronize the transmitter and receiver.The WRELAX (Weighted Fourier transform and RELAX) algorithm is used to obtain the high resolution estimation of multipath time delay.To examine the feasibility of the presented method and the accuracy and precision of the developed system,a series of sea trials are conducted in the southwest coast area of Dalian City,north of the Yellow Sea.The experimental results are compared with that using high-resolution dual echo sounder HydroBoxTM,and the uncertainty is smaller than ±0.06 m.Compared with the existing means for measuring the silt thickness,the present method is innovative,and the system is stable,efficient and provides a better real-time performance.It especially suits monitoring the narrow channel with rapid changes of siltation.  相似文献   

2.
Achieving reliable underwater communication in shallow water is a difficult task because of the random time-varying nature of multipath propagation. When the product of Doppler-related signal bandwidth spread and multipath-related time spread of the channel is larger than one, some types of adaptive signal processing may not work very well. In this paper, various methods of coherent space-time processing are compared for a condition of a marginally overspread channel operating at 50 kHz. Various combinations of suboptimal spatially adaptive and time adaptive methods are considered. The coherent path beamformer (CPB) and recursive least squares (RLS) adaptive beamformer, both in combination with RLS time filtering, are analyzed. Also considered in the analysis is the combined RLS space-time optimal adaptive processor. Many experiments using broad-band phase-shift-keyed transmissions in shallow water have been conducted to provide data for testing these various processing methods. Because of the rapid time variation of the multipath, the product of bandwidth spread and time spread at this test site approached unity. In this environment, a suboptimal approach consisting of the adaptive beamformer followed by RLS equalization reduced reverberation and transmission errors  相似文献   

3.
Underwater acoustic transient signals are generated mechanically at known positions along a wharf. These signals are received by a wide aperture planar array of four underwater acoustic sensors, whose positions relative to the wharf are unknown. A method is described that enables the positions of the sensors to be estimated from accurate differential time-of-arrival measurements (with 0.1 /spl mu/s precision) as the signal wavefronts traverse the array. A comparison of the estimated positions with the nominal positions of the first three sensors, which form a 20-m-wide aperture horizontal line array, reveals a 2-cm displacement of the middle sensor from the line array axis. This slight bowing of the line array results in overranging (bias error of 3%) when the wavefront curvature method is used with the nominal collinear sensor positions to locate a static source of active sonar transmissions at a range of 59.2 m. The use of the spherical intersection method coupled with the estimated sensor positions of the line array provides an order of magnitude improvement in the range estimate (within 0.3% of the actual value). However, systematic ranging errors are observed when the sound propagation medium becomes nonstationary. Next, the differences in the arrival times of the direct path and boundary-reflected path signals at the middle sensor of the wide aperture line array are estimated using the differential phase residue of the analytic signal at the sensor output. These multipath delays are used to estimate the range and depth of the source. Although the average value of the multipath range estimates is within 0.5% of the actual value, the variance of the range estimates is 50 times larger when compared with the results of the spherical intersection and wavefront curvature methods. The multipath delay data are also processed to provide a reliable estimate of the temporal variation in the water depth enabling the tidal variation to be observed.  相似文献   

4.
The Herault-Jutten network has been used to separate independent sound sources that have been linearly mixed. The problem of separating a mixture of several independent signals in free-field conditions or a signal and echoes in confined spaces is compounded by propagation time delays between the source(s) and the microphones because the conventional Herault-Jutten network cannot tolerate time delays. In this paper, we combine a symmetrically balanced beamforming array with the conventional Herault-Jutten network. The resulting system can adaptively separate signals that include delays introduced by the propagation medium. The proposed algorithm has been simulated in digital communication multipath channels where intersymbol interference exists. The simulation results show two clear advantages of the proposed method over the conventional adaptive equalization: (1) there is no penalty for very long impulse responses caused by long delays, and (2) no training signals are needed for equalization. The design of a multibeamformer to handle the source separation of multiple broad-band signals is also presented  相似文献   

5.
Estimation of Rapidly Time-Varying Sparse Channels   总被引:2,自引:0,他引:2  
The estimation of sparse shallow-water acoustic communication channels and the impact of estimation performance on the equalization of phase coherent communication signals are investigated. Given sufficiently wide transmission bandwidth, the impulse response of the shallow-water acoustic channel is often sparse as the multipath arrivals become resolvable. In the presence of significant surface waves, the multipath arrivals associated with surface scattering fluctuate rapidly over time, in the sense that the complex gain, the arrival time, and the Dopplers of each arrival all change dynamically. A sparse channel estimation technique is developed based on the delay-Doppler-spread function representation of the channel. The delay-Doppler-spread function may be considered as a first-order approximation to the rapidly time-varying channel in which each channel component is associated with Doppler shifts that are assumed constant over an averaging interval. The sparse structure of the delay-Doppler-spread function is then exploited by sequentially choosing the dominant components that minimize a least squares error. The advantage of this approach is that it captures both the channel structure as well as its dynamics without the need of explicit dynamic channel modeling. As the symbols are populated with the sample Dopplers, the increase in complexity depends on the channel Doppler spread and can be significant for a severely Doppler-spread channel. Comparison is made between nonsparse recursive least squares (RLS) channel estimation, sparse channel impulse response estimation, and estimation using the proposed approach. The results are demonstrated using experimental data. In training mode, the proposed approach shows a 3-dB reduction in signal prediction error. In decision-directed mode, it improves significantly the robustness of the performance of the channel-estimate-based equalizer against rapid channel fluctuations.  相似文献   

6.
被动声纳中,目标定位的测定只能利用目标声源发出的信号或噪声,爆炸声源与常用的电声式脉冲声纳发射器比较起来,具有某些明显的优点,可代替发射换能器来实现距离和方位的测定.针对水下爆炸声源,采用三元标量阵进行研究,通过确定信号到达各个阵元的时延信息来测定其目标方位和距离,并利用内插方法提高定位精度.实验结果表明,不同位置处爆...  相似文献   

7.
High-throughout multiple-access communication networks are being considered for use in underwater acoustic channels. Bandwidth limitations of underwater acoustic channels require receivers to process broad-band communications signals in the presence of several active users. To deal with the resulting multiple-access interference in addition to high intersymbol interference, the spatial variability of ocean multipath is exploited in a multichannel multiuser receiver. Two configurations of such a receiver, a centralized and a decentralized one, are presented in fully adaptive modes of operations. While greatly reducing intersymbol and multiple-access interference, spatial diversity implies high increase in adaptive multiuser receiver complexity. To reduce the complexity of the optimal multichannel combiner, spatial structure of multipath is exploited. The complexity of resulting adaptive decentralized multichannel multiuser receiver is reduced at almost no cost in performance. Comparison of proposed multichannel receivers in an experimental shallow water channel demonstrates superior performance of spatial signal combining. The use of multiple input channels is shown to provide high level of tolerance for the near-far effect in both centralized and decentralized receivers. Decentralized receiver with reduced-complexity combining is found to satisfy the performance/complexity trade-off required for practical receiver realization in shallow water networks  相似文献   

8.
A blind estimator of the ocean acoustic channel impulse response envelope is presented. The signal model is characterized by a deterministic multipath channel excited by a highly nonstationary deterministic source signal. The time-frequency (TF) representation of the received signal allows for the separation between the channel and the source signal. The proposed estimator proceeds in two steps: First, the unstable initial arrivals allow for the estimation of the source signal instantaneous frequency (IF) by maximization of the radially Gaussian kernel distribution; then, the Wigner-Ville distribution (WV) is sequentially windowed and integrated, where the window is defined by the previously estimated IF. The integral gives the channel impulse response envelope, which turns to be an approximation to the blind conventional matched filter (MF). The blind channel estimator (CE) is applicable upon the following conditions: that the multipath channel contains at least one dominant arrival well separated from the others, and that the IF of the source signal is a one-to-one function. Results obtained on real data from the INternal TIde Measurements with Acoustic Tomography Experiments (INTIMATE'96), where the acoustic channel was driven by an linear frequency modulation signal, show that the channel's envelope detailed structure could be accurately and consistently recovered, with the correlation of the estimates ranging from 0.796 to 0.973, as compared to the MF result  相似文献   

9.
Underwater acoustic communication in the multipath environment encountered in shallow water is restricted mostly by signal fading. It degrades the signal detection and time synchronization required for reliable acoustic communication. An approach to time synchronization and to the frequency diversity method is presented. A communication algorithm for obtaining a reliable acoustic underwater link, and offering an easy-to-implement decoding scheme is introduced, and system realization is described  相似文献   

10.
The effects of both small perturbations and large deformations to the array's shape on both conventional and adaptive beamformers are shown for two frequencies: the spatial Nyquist frequency (or design frequency) of the array and a frequency about three times greater. Large shape deformations lead to a decrease in the conventional beamformer's output power for a beam steered in the direction of the signal source, together with an increase in the sidelobe levels (or secondary maxima), while small perturbations in the array shape have little effect. Signal suppression is observed to be far greater for the adaptive beamformer because it is very sensitive to system errors. The imposition of a weight norm constraint on the adaptive beamformer reduces the signal suppression only for small shape perturbations array shape estimation techniques are needed to reduce signal suppression for large shape deformations. The adverse effects of a nonlinear array shape on both conventional and adaptive beamforming are shown to be substantially reduced by applying techniques that estimate the coordinates of the hydrophones prior to beamforming  相似文献   

11.
Various approaches to the beamforming of data from large aperture vertical line arrays are investigated. Attention is focused on the conventional beamforming problem where the angular power spectrum is estimated, in this case by the adaptive minimum variance processor. The data to be processed are 200 Hz CW transmissions collected at sea by a 900 m vertical line array with 120 equally spaced sensors. Correlated multipath arrivals result in signal cancellation for the adaptive processor, and spatial smoothing techniques must be used prior to beamforming. The processing of subapertures is proposed. Full aperture and subaperture processing techniques are used on the 200 Hz data. Multipath arrivals are found to illuminate only parts of the array, thus indicating that the wavefield can be highly inhomogeneous with depth  相似文献   

12.
In this study, the quality control of tide gauge observations is investigated by two methods of least square (LS-HE) and total least square harmonic estimation (TLS-HE). Particularly, it is shown how to deal with unexpected anomalies, including outliers, offset and gap in the time series of sea level height. To do so, at first the time series is constructed and then a method based on variance threshold is used to eliminate the possible outliers in the observations. Subsequently, a noise assessment algorithm is implemented and the signal is processed to find the possible times of offsets and to eliminate their corresponding observations from the time series. Finally, the signal is checked to find the periods of gap within the time series and then the gap area is predicted with correct observations. Gap filling analysis is performed in two contexts. In the first, only the significant frequencies of tide are considered in the modelling procedure, while in the second, all possible frequencies according to the period of observations are included. Our results show that for modelling and gap filling, the TLS-HE method has a better performance in a comparison with LS-HE method. Although, for offset and outlier detections, the LS-HE is recommended. It also indicates that the TLS-HE method provides a regular solution for gap filling analysis while LS-HE method needs a regularization scheme for which LSQR regularization is used.  相似文献   

13.
This paper presents the result of a first attempt to achieve a vertical synthetic aperture in the ocean for SOFAR multipath identification. The experiment was conducted during the deployment of a tomographic array in the Mediterranean Sea. Drifting the hydrophone up or down from a ship while listening to the transmitted signal created a powerful synthetic aperture at 400 Hz. Wide-band phase-coded signals, typically used in ocean tomography, were found suitable for this application. The displacement length was 100 m and the hydrophone velocity 1 m/s. The obtained resolution of 1° enabled all the rays in the tested middle range configuration to be resolved and identified. Most of them could not have been resolved with a static hydrophone. Several Doppler processing methods are presented. The narrowband approximation leading to fast algorithms is discussed. Phase time series of individual paths obtained with an array-like wave separation method show that the phase coherence of the different multipaths is nearly perfect. An angle/velocity calibration method and a first rough inversion are finally presented  相似文献   

14.
High-speed phase coherent communications in the ocean channel are made difficult by the combined effects of large Doppler fluctuations and extended, time-varying multipath. In order to account for these effects, we consider a receiver which performs optimal phase synchronization and channel equalization jointly. Since the intersymbol interference in some underwater acoustic channels spans several tens of symbol intervals, making the optimal maximum-likelihood receiver unacceptably complex, we use a suboptimal, but low complexity, decision feedback equalizer. The mean squared error multiparameter optimization results in an adaptive algorithm which is a combination of recursive least squares and second-order digital phase and delay-locked loops. The use of a fractionally spaced equalizer eliminates the need for explicit symbol delay tracking. The proposed algorithm is applied to experimental data from three types of underwater acoustic channels: long-range deep water, long-range shallow water, and short-range shallow water channels. The modulation techniques used are 4- and 8-PSK. The results indicate the feasibility of achieving power-efficient communications in these channels and demonstrate the ability to coherently combine multiple arrivals, thus exploiting the diversity inherent in multipath propagation  相似文献   

15.
Spatial diversity equalization applied to underwater communications   总被引:1,自引:0,他引:1  
Underwater acoustic digital communication is difficult because of the nature of the fading multipath channels. Digital signal processing, such as adaptive equalization, is known to greatly improve the communication data rate by limiting intersymbol interference (ISI). However, existing underwater acoustic equalization studies are limited to single-channel techniques, and spatial diversity processing is limited to selection or combining. In this paper, we design minimum mean-square error (MMSE) equalizers jointly among all spatial diversity channels. We call this spatial diversity equalization (SDE). Results are based on a very sparse vertical array in a midrange underwater acoustic channel. We study the effect of element number and placement, the length of the equalization filters, and linear feedforward versus nonlinear decision feedback algorithms. A suboptimum equalizer combiner (EC) is studied to alleviate the computational intensity of JCE. We first design the system for a known acoustic channel; later, some results are verified using adaptive algorithms. Results are presented both in terms of the mean-square error (MSE) and the probability of a symbol error. The latter is important as it is the ultimate interest for a digital communication system. We found that system performance improves rapidly with an increase in the number of spatial channels  相似文献   

16.
传统罗兰C接收机为避免天波信号干扰,采用固定的采样时刻,以致接收机信噪比下降,直接影响了接收机的整体工作性能.分析了三种测量天波时间延迟的现代信号处理方法,并进行了实验仿真分析比对,验证了该方法对于测量天波延迟地波时间的准确性和有效性.  相似文献   

17.
In the analysis of crosshole tomography data, the first step is to estimate the arrival time and amplitude of the multi-path arrivals which comprise the received signal. Normally algorithms such as matched filter are used to determine the arrival times. However, when the bandwidth of the signal is small, this method cannot resolve closely spaced arrivals. We, therefore, investigate the performance of a simulated annealing algorithm in estimating the amplitude scaling factors and delay times of the separate arrivals in a signal composed of closely spaced arrivals with added noise. The algorithm is applied to field data collected during a crosshole tomography experiment conducted in sea ice  相似文献   

18.
A design procedure for an amplitude-modulated and nonlinear frequency-modulated (AM-NLFM) signal is introduced. The designed signal can drive a given transducer to its peak power to produce a sound pressure waveform into the water with a desired power spectrum and maximum possible energy. The signal can be formed either in the time domain or in the frequency domain. The frequency domain approach gives an output power spectrum precisely identical to a preferred shape. Therefore, the sidelobe levels after matched filtering are not raised by unwanted spectral magnitude ripples which exist when a time domain method is adopted. The absence of spectral ripples is desirable for applications requiring long range transmission and good multipath discrimination capability. An acceptable tradeoff between time resolution and sidelobe levels is achieved by properly choosing the desired power spectral shape. As the time resolution is usually the most critical specification for precision travel-time measurements, it is shown that by sacrificing some of the transducer's output power capability, a waveform with a considerably wider bandwidth can be transmitted, resulting in a significantly enhanced time resolution. A quasi-steady-state (QSS) approximation is used in the signal design, leading to a manageable and intuitive design procedure  相似文献   

19.
The passive-range estimation technique using a single focused beamformer has been studied under the sea. However, there are not many more methods in the case of closed multisource environments. In this paper, we propose the technique using dual focused beamformers to estimate the ranges of a closed multisource, and compare the proposed technique with the previous method. The proposed method is verified via computer simulation under the simplified multipath underwater channel model.  相似文献   

20.
In active sonar and in passive sonar localization, time delay is a fundamental parameter whose extraction is vital to the sonar function. The underlying time delay parameter (or parameter vector) contains information about the acoustic source (or reflector) as seen through the ocean at a receiver. The ocean effects require sonar adaptation. A tutorial review of ocean effects in time delay estimation is provided, with references to benchmark work. It covers coherence, time-delay estimation, localization, time-varying time delay estimation, the complexity of the ocean environment, and depth estimation using mode matching  相似文献   

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