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.     

Digital underwater acoustic voice communications

This paper describes the design of an underwater acoustic diver communication system controlled by a digital signal processor. The speech signal transmission rate is compressed by using linear predictive coding (LPC) and the extracted parameters are transmitted through the water to a synchronized receiver by employing digital pulse position modulation (DPPM). The pulse position in each time frame is estimated by an energy detection and decision algorithm which enables the received LPC parameters to be recovered and used to synthesize the speech signal     

Low complexity adaptive equalizers for underwater acoustic communications

Interference signals due to scattering from surface and reflecting from bottom is one of the most important problems of reliable communications in shallow water channels. To solve this problem, one of the best suggested ways is to use adaptive equalizers. Convergence rate and misadjustment error in adaptive algorithms play important roles in adaptive equalizer performance. In this paper, affine projection algorithm （APA）, selective regressor APA（SR-APA）, family of selective partial update （SPU） algorithms, family of set-membership （SM） algorithms and selective partial update selective regressor APA （SPU-SR-APA） are compared with conventional algorithms such as the least mean square （LMS） in underwater acoustic communications. We apply experimental data from the Strait of Hormuz for demonstrating the efficiency of the proposed methods over shallow water channel. We observe that the values of the steady-state mean square error （MSE） of SR-APA, SPU-APA0 SPU-normalized least mean square （SPU-NLMS）, SPU-SR-APA0 SM-APA and SM-NLMS algorithms decrease in comparison with the LMS algorithm. Also these algorithms have better convergence rates than LMS type algorithm.     

Recent advances in high-speed underwater acoustic communications

In recent years, underwater acoustic (UWA) communications have received much attention as their applications have begun to shift from military toward commercial. Digital communications through UWA channels differ substantially from those in other media, such as radio channels, due to severe signal degradations caused by multipath propagation and high temporal and spatial variability of the channel conditions. The design of underwater acoustic communication systems has until recently relied on the use of noncoherent modulation techniques. However, to achieve high data rates on the severely band-limited UWA channels, bandwidth-efficient modulation techniques must be considered, together with array processing for exploitation of spatial multipath diversity. The new generation of underwater communication systems, employing phase-coherent modulation techniques, has a potential of achieving at least an order of magnitude increase in data throughput. The emerging communication scenario in which the modern underwater acoustic systems mill operate is that of an underwater network consisting of stationary and mobile nodes. Current research focuses on the development of efficient signal processing algorithms, multiuser communications in the presence of interference, and design of efficient modulation and coding schemes. This paper presents a review of recent results and research problems in high-speed underwater acoustic communications, focusing on the bandwidth-efficient phase-coherent methods. Experimental results are included to illustrate the state-of-the-art coherent detection of digital signals transmitted at 30 and 40 kb/s through a rapidly varying one-mile shallow water channel     

Temporal resolutions of time-reversal and passive-phase conjugation for underwater acoustic communications

In this paper, we study the temporal resolution of a time-reversal or passive-phase conjugation process as applied to underwater acoustic communications. Specifically, we address 1) the time resolution or the pulse width of a back-propagated time-compressed pulse as compared with the original transmitted pulse; 2) the effectiveness of temporal focusing as measured by the peak-to-sidelobe ratio of the back-propagated or phase-conjugated pulse (both pulse elongation and sidelobe leakages are causes of intersymbol interference and bit errors for communications); 3) the duration of temporal focusing or the temporal coherence time of the underwater acoustic channel; and 4) the stability of temporal focusing as measured by the phase fluctuations of successive pulses (symbols). Binary phase-shift keying signals collected at sea from a fixed source to a fixed receiver are used to extract the above four parameters and are compared with simulated results. Mid-frequency (3-4-kHz) data were collected in a dynamic shallow-water environment, exhibiting high temporal fluctuations over a scale of minutes. Despite this, the channel is found to be highly coherent over a length of 17 s. As a result, only one probe signal is used for 17 s of data. The bit error rate and variance of the symbol phase fluctuations are measured as a function of the number of receivers. They are of the same order as that calculated from the simulated data. The agreement suggests that these two quantities could be modeled for a communication channel with high coherence time. The phase variance can be used to determine the maximum data rate for a phase-shift keying signal for a given signal bandwidth and a given number of receivers.     

北极冰下水声通信中改进的最小均方/四次方直接自适应均衡器

本文提出了一种适用于北极冰下水声通信的最小均方/四次方直接自适应均衡器（LMS/F-DAE）。它能处理基带复信号,与LMS相比,具有更好的均衡效果。考虑到均衡器的稀疏特性,在其代价函数中加入自适应范数（AN）作为约束。它能根据均衡器系数的大小自适应变化：对于小系数,此约束项存在以加快收敛速度;对于大系数,此约束项不存在以减小均衡误差。利用第九次中国北极科学考察得到的实验数据验证AN-LMS/F-DAE的性能。结果表明,与传统的LMS/F-DAE相比,AN-LMS/F-DAE能提升均衡器的稀疏性且均衡性能更优。     

Phase-coherent digital communications for underwater acousticchannels

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     

Signal detection for communications in the underwater acousticenvironment

Signal detection is a critical stage in the implementation of any effective communications system. The underwater acoustic environment, particularly in the presence of underwater vehicles, presents significant challenges to reliable detection without excessive false alarms. While there is often sufficient signal-to-noise ratio with respect to stationary broad-band background noise to permit reliable operation, the presence of strong event-like interference signals such as narrow-band signals and impulsive broad-band signals complicates the detection problem significantly. Frequency-hopped signals interleaved with quiescent bands are proposed as the basis of a robust detection system. These signals also make robust detection possible in a multi-access communications system. Two new detection algorithms that exploit the particular structure of these frequency-hopped signals are developed and their performance is analyzed. This analysis uses a modification of the doubly noncentral F-distribution that has not been used previously for such analysis. This distribution makes possible the direct calculation of probabilities of detection and false-alarm under interference and signal scenarios that cannot be analyzed using the better known noncentral F-distribution. With this analysis, the two developed detectors are shown to offer superior performance to that of either the CFAR detector or the binary data sequence detector. Experimental data confirms the theoretically derived results     

Spatial diversity processing for underwater acoustic telemetry

A large increase in the reliability of shipboard or stationary underwater acoustic telemetry systems is achievable by using spatially distributed receivers with aperture sizes from 0.35 to 20 m. Output from each receiver is assigned a quality measure based on the estimated error rate, and the data, weighted by the quality measure, are combined and decoded. The quality measure is derived from a Viterbi error-correction decoder operating on each receiver and is shown to perform reliability in a variety of non-Gaussian noise and jamming environments and reduce to the traditional optimal diversity system in a Gaussian environment. The dynamics of the quality estimator allow operation in the presence of high-power impulsive interference by exploiting the signal and noise differential travel times to individual sensors. The spatial coherence structure of the shallow water acoustic channel shows relatively low signal coherence at separations as short as 0.35 m. Increasing receiver spacing beyond 5 m offers additional benefits in the presence of impulsive noise and larger-scale inhomogeneities in the acoustic field. A number of data transmission experiments were carried out to demonstrate system performance in realistic underwater environments     

水下光学无线通信的海水信道特性研究

提出了将具有比声学高达几倍的数据通信速率,良好的安全性和隐蔽性的光学无线通信技术应用于海军、海洋科学研究和水下工程等领域,实现高速率的水下无线通信技术实现海量数据的信息交换.基于生物光学特性的水下光学信道模型,建立了水下光学通信系统性能分析方法.并对基于发光二极管(LED)的水下光学无线通信系统进行了仿真,其结果表明所建立的方法可以进行各种海域水质环境的模拟,便于时水下光学无线通信系统性能进行预测评估,为水下光学无线通信系统的设计方案的评估提供了依据.     

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     

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.     

Chiral composites as underwater acoustic attenuators

A plane-wave propagation in an elastic matrix containing the structural chiral microstructures is employed to model the dynamic response of a particles-mixture composite. Two nondispersive longitudinal wavenumbers and four dispersive circularly polarized transverse wavenumbers result from the dispersion equation of the so-called effective chiral (isotropic, noncentrosymmetric) composite. Our previous research indicated both that two transition frequencies divide the frequency spectrum of the transverse wavenumbers into three varying groups and that the four transverse modes can be distinguished only in a specified frequency range. This study illustrates the reflected and transmitted characteristics at a fluid-chiral interface at certain frequencies. The reflected and transmitted fields at the fluid-isotropic interface are solved to depict the effects of the chirality. The chiral material should instigate a reducible reflected plane wave and may be used as an anechoic coating to “absorb” sound underwater, due to the mode conversion of the chirality in the chiral medium     

An integrated sensory-intelligent system for underwater acoustic signal-processing applications

A generic integrated sensory-intelligent system (ISIS) is developed for underwater acoustic signal-processing applications. ISIS constantly monitors the current acoustic channel conditions and smoothly integrates the outputs of the most appropriate signal-processing procedures or algorithms available to it for those conditions. The system is based on a generalization of a tuneable approximate piecewise linear (TAPL) model derived from the modified probabilistic neural network (MPNN). This model was designed to seamlessly integrate a set of local linear signal-processing algorithms within a given multidimensional data space. Depending on the input signal distortions, which are determined by environmental effects, ISIS automatically weighs and adds the outputs from a set of processing algorithms working in parallel. The weighting is related to the "closeness" of each algorithm to the sensed input signal characteristics or some other measured environmental state. A single tuning parameter is used to smoothly and seamlessly select appropriately among the parallel processing algorithm outputs. A very small tuning-parameter value selects the closest most appropriate algorithm output. At the other extreme, a fixed weighted average of all the algorithm outputs is produced with a very large value. Otherwise, a dynamic weighed average of all algorithm outputs is achieved with values in between. Some features and benefits of ISIS are demonstrated with an illustrative linear sweep chirp signal-detector estimation problem characterized by extremely variable Doppler conditions.     

Performance limitations in underwater acoustic telemetry

Performance limitations in digital acoustic telemetry are addressed. Increases in computational capabilities have led to a number of complex but practical solutions aimed at increasing the reliability of acoustic data links. These solutions range from ocean-basin scale data telemetry to video-image transmission at a few hundred yards' distance. The opportunity to implement highly complex tasks in real time on modest hardware is a common factor. The data rates range from 1 to 500 kb/s and are much slower than satellite channels, while acceptable system complexity is higher than virtually any other channel with comparable data throughput. The basic performance bounds are the channel phase stability, available bandwidth, and the channel impulse response fluctuation rate. Phase stability is of particular concern for long-range telemetry, channel fluctuation characteristics drive equalizer, and synchronizer design; the bandwidth limitation is a direct constraint on data rate for a given signaling method     

一种改进的水下声学定位算法

通过分析讨论泰勒级数展开法和Chan算法的优缺点,提出了一种先由Chan算法解算求得初始参考值,再由泰勒级数展开法在该初始值处展开进行迭代求解的协同定位算法。通过实验,将本方法与传统定位算法(泰勒法)、Chan算法进行对比。结果表明,该方法无需额外提供初始值,解算出较好的结果的同时且能保持良好的时间效率。通过此方法,在实际生产作业中可节约定位时间,改善定位精度。     

Algorithms for joint channel estimation and datarecovery-application to equalization in underwater communications

Methods for joint ocean-channel estimation and data recovery are derived using an optimal, maximum likelihood (ML) estimation criterion. The resulting ML problems may be complex, thus iterative algorithms are used, e.g. the expectation-maximization (EM) algorithm. The different methods correspond to different assumptions about the ocean channel. The theoretical derivation of these methods as well as preliminary results on simulated ocean data experiments are presented     

A high-resolution pulse-Doppler underwater acoustic navigation system

A high-resolution underwater acoustic pulse-Doppler navigation system has been developed and tested at sea. The system provides continuous, highly accurate tracking of underwater and ocean-surface platforms in a fixed 50-km²navigation net. Three reference buoys, moored 20 m from the ocean bottom, provide the navigation net used by shipboard processing equipment. Each reference buoy contains an acoustic transponder, used to obtain the acoustic travel times from the transponder to the platform, and a continuous-tone beacon, used to obtain the Doppler shift due to platform motion. The system is capable of determining the position of a platform with respect to the reference net with an error of 2-3 m. The relative position of the platform on a fix-to-fix basis can be determined within several centimeters over short time intervals (approx 10min).