Multichannel processing of broad-band multiuser communicationsignals in shallow water acoustic channels

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     

Improvement of Time-Reversal Communications Using Adaptive Channel Equalizers

The spatial and temporal focusing properties of time-reversal methods can be exploited for undersea acoustic communications. Spatial focusing mitigates channel fading and produces a high signal-to-noise ratio (SNR) at the intended receivers along with a low probability of interception elsewhere. While temporal focusing (compression) reduces significantly intersymbol interference (ISI), there always is some residual ISI depending upon the number of transmitters, their spatial distribution (spatial diversity), and the complexity of the channel. Moreover, a slight change in the environment over the two-way propagation interval introduces additional ISI. Using multilevel quadrature amplitude modulation (M-QAM) in shallow water, we demonstrate that the performance of time-reversal communications can be improved significantly by cascading the received time series with an adaptive channel equalizer to remove the residual ISI     

Spatial diversity equalization applied to underwater communications

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     

Differences between passive-phase conjugation and decision-feedback equalizer for underwater acoustic communications

Passive-phase conjugation (PPC) uses passive time reversal to remove intersymbol interferences (ISIs) for acoustic communications in a multipath environment. It is based on the theory of signal propagation in a waveguide, which says that the Green's function (or the impulse-response function) convolved with its time-reversed conjugate, summed over a (large-aperture) vertical array of receivers (denoted as the Q function) is approximately a delta function in space and time. A decision feedback equalizer (DFE) uses a nonlinear filter to remove ISI based on the minimum mean-square errors (mmse) between the estimated symbols and the true (or decision) symbols. These two approaches are motivated by different principles. In this paper, we analyze both using a common framework. We note the commonality and differences, and pros and cons, between the two methods and compare their performance in realistic ocean environments, using simulated and at-sea data. The performance measures are mean-square error (mse), output signal-to-noise ratio (SNR), and bit-error rate (BER) as a function of the number of receivers. For a small number of receivers, the DFE outperforms PPC in all measures. The reason for poor PPC performance is that, for a small number of receivers, the Q function has nonnegligible sidelobes, resulting in nonzero ISI. As the number of receivers increases, the BER for both processors approaches zero, but at a different rate. The modeled performance differences (in mse and SNR) between PPC and DFE are in general agreement with the measured values from at-sea data, providing a basis for performance prediction.     

Spatial smoothing and minimum variance beamforming on data fromlarge aperture vertical line arrays

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     

一个水声扩频通信系统设计与实现

严重的多途衰落、多普勒频偏是水声通信中引起误码的主要原因。低功耗、远距离、高隐蔽性、低信噪比检测、高可靠性的数据传输是水声通信的一个发展方向。设计并实现了一个水声扩频通信系统,有效地解决了以上问题,并采用快速相关算法,代替传统的矢量与矩阵相乘运算,极大地减少了程序的运行量,从而实时地处理接收信号。通过湖试和海试,验证了此通信系统的优良性能。     

Adaptive port-starboard beamforming of triplet sonar arrays

For a low-frequency active sonar (LFAS) with a triplet receiver array, it is not clear in advance which signal processing techniques optimize its performance. Here, several advanced beamformers are analyzed theoretically, and the results are compared to experimental data obtained in sea trials. Triplet arrays are single line arrays with three hydrophones on a circular section of the array. The triplet structure provides the ability to solve the notorious port-starboard (PS) ambiguity problem of ordinary single-array receivers. More importantly, the PS rejection can be so strong that it allows to unmask targets in the presence of strong coastal reverberation or traffic noise. The theoretical and experimental performance of triplet array beamformers is determined in terms of two performance indicators: array gain and PS rejection. Results are obtained under several typical acoustic environments: sea noise, flow noise, coastal reverberation, and mixtures of these. A new algorithm for (beam space) adaptive triplet beamforming is implemented and tuned. Its results are compared to those of other triplet beamforming techniques (optimum and cardioid beamforming). These beamformers optimize for only one performance indicator, whereas in theory, the adaptive beamformer gives the best overall performance (in any given environment). The different beamformers are applied to data obtained with an LFAS at sea. Analysis shows that adaptive triplet beamforming outperforms conventional beamforming algorithms. Adaptive triplet beamforming provides strong PS rejection, allowing the unmasking of targets in the presence of strong directional reverberation (e.g., from a coast) and at the same time provides positive array gain in most environments.     

基于二阶锥规划的空域维纳滤波

提出了基于二阶锥规划的空域维纳滤波。基于噪声与阵列流型不相关的特性,将阵列的方向向量作为期望通过二阶锥规划方法对阵列快拍进行维纳滤波。利用维纳滤波抑制噪声的能力,提高阵列的信号检测和方位估计性能。通过仿真研究,验证了单目标时,其低信噪比下的信号检测性能优于Bartlett波束形成,其方位估计的信噪比门限要低于Bartlett波束形成和MUSIC算法,因而是一种优良的波束形成器。该波束形成器用于多波束测深仪和侧扫声纳等海洋测绘设备的阵列,可以有效的提高设备的作用距离和测量精度。     

Angular-spectral decomposition beamforming for acoustic arrays

The method of principal component beamforming described in this paper is an array data reduction method that allows one to observe the statistically uncorrelated components of wave energy arriving at an array of acoustic sensors. The method can be used to process array data so as to observe and identify the sources of noise, both environmental and self noise. After identifying the sources of noise, the method of principal components can be used to discriminate signal from noise. The method can be applied to active systems (subbottom profilers) as well as passive systems. A model of isotropic noise and incident bandlimited plane waves is used to study array resolution and bandwidth effects. Experimental data from a2 times 3planar acoustic array were used to identify sources of hydro-flow related noise in an underwater vehicle. In all cases studied, the technique provides a maximum spatial information analysis method to the observer.     

Correlation-based decision-feedback equalizer for underwater acoustic communications

The purpose of this paper is to develop a decision-feedback equalizer (DFE) using a fixed set of parameters applicable to most shallow oceans with minimal user supervision (i.e., a turn key system). This work is motivated by the superior performance [bit error rate (BER)] of the multichannel DFE compared with other methods, such as passive-phase conjugation (PPC), at the same time noting its sensitivity to different acoustic environments. The approach is to couple PPC, utilizing its adaptability to different environments, with a single-channel DFE. This coupling forms an optimal processor for acoustic communications in theory, but it has never been implemented in practice. By coupling with DFE, the method achieves the same spatial diversity as conventional multichannel DFE, without requiring a large number of receivers as does PPC. The correlation-based DFE in terms of the autocorrelation functions of the channel impulse responses summed over the receiver channels (the Q function) is derived. This paper shows in terms of waveguide physics, further supported by real data, the many desirable features of the Q function that suggest, given adequate sampling of the water column, a general applicability of the correlation-based equalizer to different environments, irrespective of the sound speed profiles, bottom properties, and source-receiver ranges/depths. This property can be expected to hold approximately for a small number of receivers with spatial diversity. This paper demonstrates the robustness of the new equalizer with moving source data despite the range change (which modifies the impulse response) and symbol phase change due to time-varying Doppler.     

Multiple terminal acoustic communications system design

A design is presented for a system providing highly reliable command and control acoustic communications between a mother ship and a number of small fast submersibles. The small submersibles may be employed for underwater mining, exploration, bottom mapping, or military surveillance. Modulation and coding design is presented; the techniques discussed provide multiple protection against multipath and fading, high reliability, acceptable transmitted signal total time duration, simplicity, and economy. The required decision point signal-to-noise ratio (SNR) for Rayleigh fading conditions is derived for the modulation and coding design. Particular attention is paid in the receive signal processing to the Doppler (relative velocity) and Doppler variation (relative acceleration) problems inherent in a scenario with mobile endpoints. A Figure-of-Merit (FOM) calculation is provided for typical geometrical and environmental parameters. It is shown for a realistic source level that the required SNR can be achieved at long range with considerable endpoint relative motion.     

Searching for seafloor massive sulfides: a quantitative review of high-resolution methods in deep sea sonar bathymetry for mining applications

Seafloor massive sulphides are deep sea mineral deposits currently being examined as a potential mining resource. Conventional sonar bathymetry products gathered by sea surface platforms do not achieve adequate spatial resolution to detect these resources. High-resolution beamforming methods (such as multiple signal classification and estimation of signal parameters via rotational invariance techniques) improve the resolution of sonar bathymetry. We perform a quantitative review of these high-resolution methods using a novel simulator, showing results in the absence of platform motion for a single ping cycle. It was found that high-resolution methods achieved greater bathymetric accuracy and higher resolution than conventional beamforming and that these methods may be adequate for this style of marine exploration. These methods were also robust in the presence of unwanted persistent signals and low signal to noise ratios.     

High-frequency ambient noise and its impact on underwater tracking ranges

A theoretical model for the vertical directionality and depth dependence of high frequency (8 to 50 kHz) ambient noise in the deep ocean is developed. The anisotropic noise field at a variety of depths and frequencies is evaluated and displayed. It was found that at high frequencies and deep depths, a bottom-mounted hydrophone receives the maximum noise energy from overhead rather than from the horizontal. This leads to the consideration of an oblate hydrophone receiving response pattern for underwater tracking ranges that would provide a constant signal-to-noise ratio (SNR) for an acoustic source located anywhere in a circular area centered above the hydrophone. Two of the desirable characteristics of this type of pattern are the increase in receiving range of a bottom-mounted sensor and the decrease of the dynamic range of signals that a signal processor must handle.     

A method minimizing the correlation properties of a noise field andimproving detection of signals embedded in anisotropic noise

This paper deals with the development of a processing technique that improves the signal-to-noise ratio (SNR) at the single sensor for a received signal that is embedded in a partially correlated noise field. The approach of this study is unique in that the noise is treated as being non-white and partially correlated. The concept of the proposed development is based on the time interval over which the temporal coherence or correlation properties of a noise field are defined. For narrowband signals, the associated temporal coherence period is much longer than the correlation time interval of the anisotropic noise field. Thus, a coherent integration of discontinuous segments of received signals will enhance the SNR at the single sensor by lowering the correlation properties of the associated non-white noise. Reconstruction of the narrow-band signal time series, with improved SNR at the sensor will allow the use of the existing high resolution techniques to be utilized more effectively by lowering their threshold values in order to detect very weak signals. The intention here is to integrate the characteristics of the real anisotropic noise field during the preliminary processing stages of the received signals by an array of sensors. Simulations show that the proposed method can be integrated in the signal processing functionality of sonar and radar systems     

Acoustic telemetry system for underwater control

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     

High-resolution beamforming by the Wigner-Ville distribution method

The Wigner-Ville distribution (WVD) function was originally proposed by Wigner in quantum mechanics and Ville applied it for signal analysis. This method has made it possible to represent a signal's power density spectrum in the time-frequency domain as a natural extension of the Fourier transform method (FTM). Recently, it has attracted great interest for its validity to analyze time-varying signals accomplished by the development of high-speed digital signal processing, and it is used for analyzing nonstationary signals. Conventionally, a sonar beamformer is constructed using delay lines, but the development of the high-speed processor has made it possible to apply the FTM for sonar beamforming. However, the bearing resolution of the beamformer is not enough for discriminating small underwater objects on the sea bottom by this method. To solve this problem, we aim to apply the WVD method, which can represent finer structure of signals as a natural extension of the FTM, for sonar beamforming to obtain sharper beam patterns than those of the beamforming method by FTM. Simulation results by computational calculations to clarify the resolution by the WVD method, which is presented in this paper, becomes approximately twice as high as by conventional FTM. The results of an experiment at sea also show the performance of this method     

Passive sonar detection and localization by matched velocityfiltering

This paper presents a new bearings-only method of detecting and tracking low signal-to-noise ratio (SNR) wideband targets on a constant course and velocity trajectory. A track-before-detect strategy based on matched velocity filtering is adopted using spatial images constructed from a sequence of power bearing map (PBM) estimates accumulated during a track. To lower the threshold SNR for detection, a discrete bank of matched velocity filters integrates the PBM images over a range of hypothesized trajectories, such an approach eliminates the need to estimate the number of targets since signal detection is determined by comparing the output of each matched filter (MF) to a decision threshold. The distribution of the MF output is derived based on a single point target in diffuse noise assumption. Receiver operating characteristic curves show a definite detection gain under low SNR conditions for matched velocity filtering (track-before-detect) over detection from a single PBM     

Analysis of channel effects on direct-sequence and frequency-hoppedspread-spectrum acoustic communication

Multiuser underwater acoustic communication is one of the enabling technologies for the autonomous ocean-sampling network (AOSN). Multiuser communication allows vehicles, moorings, and bottom instruments to interact without human intervention to perform adaptive sampling tasks. In addition, multiuser communication may be used to send data from many autonomous users to one buoy with RF communications capability, which will then forward the information to shore. The two major signaling techniques for multiuser acoustic communication are phase-shift keying (PSK) direct-sequence spread-spectrum (DSSS) and frequency-shift keying (FSK) frequency-hopped spread-spectrum (FHSS). Selecting between these two techniques requires not only a study of their performance under multiuser conditions, but also an analysis of the impact of the underwater acoustic channel. In the case of DSSS, limitations in temporal coherence of the channel affect the maximum spreading factor, leading to situations that may be better suited to FHSS signals. Conversely, the multipath resolving properties of DSSS minimize the effects of frequency-selective fading that degrade the performance of FSK modulation. Two direct-sequence receivers potentially suitable for the underwater channel are presented. The first utilizes standard despreading followed by decision-directed gain and phase tracking. The second uses chip-rate adaptive filtering and phase tracking prior to despreading. Results from shallow water testing in two different scenarios are presented to illustrate the techniques and their performance     

Adaptive separation of mixed broadband sound sources with delays bya beamforming Herault-Jutten network

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     

The effect of phase-shifter errors on the performance of an antenna-array beamformer

The paper considers the random phase errors in the phase shifters which are used in an antenna array to steer the beam in the look direction, and analyzes the effect of these errors on the performance of the optimal processor which maximizes the output SNR by deriving the expressions for the output signal power, residual interference power, output SNR, and the array gain as a function of the variance of these errors. The paper also considers the phase quantization error which arises in the digital phase shifters and shows how the performance of the optimal processor depends on the number of bits of the digital phase shifters.