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     

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     

Equalizer for video rate transmission in multipath underwatercommunications

This paper concerns the equalization problem of an underwater high rate transmission system. Because the channel delay spread of an horizontal link is large compared with the data duration, we have developed an adaptive equalizer in order to minimize the calculus burden of the moving receiver. The performance of this equalizer on synthetic and real channels is discussed     

Single-mode excitation in the shallow-water acoustic channel usingfeedback control

The shallow-water acoustic channel supports far-field propagation in a discrete set of modes. Ocean experiments have confirmed the modal nature of acoustic propagation, but no experiment has successfully excited only one of the suite of mid-frequency trapped modes propagating in a coastal environment. The ability to excite a single mode would be a powerful tool for investigating shallow-water ocean processes. A feedback control algorithm incorporating elements of adaptive estimation, underwater acoustics, array processing, and control theory to generate a high-fidelity single mode is presented. This approach also yields a cohesive framework for evaluating the feasibility of generating a single mode with given array geometries, noise characteristics, and source power limitations. Simulations and laboratory wave guide experiments indicate the proposed algorithm holds promise for ocean experiments     

Compound compensation strategies for wireless data communicationsover the multimodal acoustic ocean waveguide

Underwater acoustic communications (UAC) at the reverberation-limited range results in severely distorted information signals. Wide-band signals are subject to both intermodal and intramodal-type of dispersions. The underwater acoustic channel impulse response and the sidelobes strongly depend on the waveguide structure and the source and receiver positions. The motion and displacement from this position, as well as other environmental variabilities impose a real-time adaptivity for the receiver operation to keep track of the fluctuations. To increase the system's reliability and data rate, there is a need to employ adaptive equalizers and diversity techniques to improve the margins against noise, and intersymbol interference (ISI). Blind adaptive equalization (BAE) is the ideal adaptive compensation when operating point-to-multipoint networks, and centralized communication systems in general. Inherent optimum multiple resonant modes within the ocean acoustic waveguide can be exploited judiciously via a new proposed parallel data multicarrier modulation (MCM) scheme by sending data over the multiple subcarriers. MCM might eventually obviate equalization which introduces higher-order computational complexity to the receiver. The above modulation eliminates multipaths and allows operation at multiples of the single-carrier transmission rate. The system's immunity to distortions such as ISI, fast fades, and impulsive noises, is increased due to incorporation of symbol guard space. Direct comparisons with single carrier schemes (such as higher-order statistics (HOS)-based equalization) are of great interest, since the proposed new receiver configuration has low-complexity to provide a compact, portable and low-power practical acoustic modem. Finally, network topology issues are considered to determine optimum network architectures for underwater acoustic LANs. A central topology (CT) supported by BAE and MCM transmission is proposed     

MFSK水声通信接收系统中的幅度均衡设计

为了抗幅度随频率的衰减，提出了一种基于多频移频健控（MFSK）调制方式的水声通信接收系统中的幅度均衡技术，介绍了构成幅度均衡电路的基本原理以及在厦门港浅海域中的实验结果，实验结果表明，该技术能有效克服上系统中的信号民幅度随率的衰减问题。使接收到的图像清晰易读。     

多模式自适应水下无线通信网络框架研究

对当前典型的水下无线通信网进行分析,针对水声、光、射频3种通信模式在水下无线通信中的优缺点,提出基于软件无线电技术的多模式自适应水下无线通信网络的概念及其框架结构,并对其中的自适应调制解调方式展开研究.结合MAC层协议,提出一种跨层的自适应调制解调解决方案,即通过收发双方的握手信息携带当前信道状态,由发射方根据握手信息,判断双方通信距离,预计信道未来状态,结合需要传输的数据量,自适应选择合适的通信模式和调制方式,并利用握手信号通知接收方,从而实现在通信网络范围内数据或指令的快速可靠传输.     

Parametric array Doppler sonar

A Doppler sonar technique capable of measuring ground speed and distance traveled at extreme ocean depths is described. The technique uses nonlinear acoustic effects to create the narrow beams required. Performance estimates are calculated and parameters for a test system chosen. Sonar Pond test results of transmitter beam patterns and source levels are given. The sea trials conducted in July and August of 1975, aboard the R. V. Sperry Star off New Providence Island, are described. Evaluation of data taken in water depths to 11 000 ft is made and substantial agreement with original estimates is demonstrated. The feasibility of an accurate, all ocean capability, dead reckoning navigator is thus confirmed.     

An ocean bottom,microprocessor based seismometer

We describe the design and construction of an ocean bottom seismometer configured as a computer, based on an Intersil IM6100 microprocessor plus appropriate peripheral devices. The sensors consist of triaxial 1 Hz seismometers and a hydrophone, each sensor channel being filtered prior to digitizing so that typical noise spectra are whitened. Digital data are recorded serially on magnetic tape. The instrument is placed on the ocean bottom by allowing it to fall freely from just below the surface. An acoustic system allows precise determination of instrument position, acoustic recall, and transmission of operational information to the surface. Release from an expendable anchor is accomplished by redundant pyrotechnic bolts which can be fired by acoustic command or by precision timers.The operational flexibility provided by the micro-computer, which executes the DEC PDP8/E instruction set, enables optimum use of the 6-hr recording capacity (at 128 samples/second/channel) in the context of the particular experiment being performed.

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时变水声信道中基于BEM的迭代信道估计技术

水声信道的多径时延扩展和时变特性对信道估计和均衡技术的研究带来了很大的挑战,同时也决定了水声信道是一种时频双扩展信道,提出一种水声OFDM通信系统中基于软信息的迭代信道估计技术,利用基于复指数基扩展模型(CE-BEM)进行信道估计。OFDM系统本身可以消除由于多径引起的符号间干扰(ISI)。基于导频的BEM信道估计,可以实现对时变信道的估计,结合基于软信息迭代的迭代均衡模块,将每次迭代生成的符号软判决信息作为辅助导频用于信道估计。同时,为了防止由于信道时变引起的信道子载波间干扰(ICI)对导频符号的影响,采用基于保护间隔的导频插入法插入导频。仿真结果显示基于BEM的软信息迭代信道估计性能较非迭代信道估计时明显提升。     

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     

Underwater acoustic channel models for 4800 b/s QPSK signals

Two computer models are presented, one for short-range and one for long-range propagation of acoustic signals through an underwater channel from a transmitter to a receiver. In the short-range model, the received signal is due to a direct path (steady component) and a random path (diffused component) that could be the result of boundary scattering. For the long-range case, the received signal is the superposition of a number of time-delayed, randomly propagated components arriving by different paths. Both models assume perfect transmitter-receiver synchronization but use realistic channel time delays. They demonstrate the time-varying characteristics of underwater acoustic channels and are used in simulations to evaluate the performance of the detection technique     

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     

水声信号时频分析方法比较及应用研究

浅海中的宽带水声信号传播呈现出频散的特点,通过高分辨率的时频分析方法可以刻画频散曲线。通过数值仿真和实验数据处理,对比分析几类常用的时频分析方法在提取宽带声信号频散曲线方面的性能。结果表明:STFT时频局部化精度不够高;在较强频散的情况下,DSTFT时频分辨率较高。WVD时频聚集性最好,但是有严重的交叉项干扰;固定核函数的CWD较好地抑制交叉项,时频分辨率虽优于STFT,但弱化了时频聚集性;AOK时频分布采用自适应高斯核函数,在抑制交叉项的同时,时频聚集性较好,有望较好地用于提取信号频散曲线。     

Blind equalization based on RLS algorithm using adaptive forgetting factor for underwater acoustic channel

Blind equalization based on adaptive forgetting factor, recursive least squares （RLS） with constant modulus algorithm （CMA）, is investigated. The cost function of CMA is simplified to meet the second norm form to ensure the stability of RLS-CMA, and thus an improved RLS-CMA （RLS-SCMA） is established. To further improve its performance, a new adaptive forgetting factor RLS-SCMA （ARLS-SCMA） is proposed. In ARLS-SCMA, the forgetting factor varies with the output error of the blind equalizer during the iterative process, which leads to a faster convergence rate and a smaller steady-state error. The simulation results prove the effectiveness under the condition of the underwater acoustic channel.     

Estimation of Rapidly Time-Varying Sparse Channels

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.     

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。     

海底底质声学性质原位测量技术研究

系统介绍了海底底质声学性质原位测量技术,提出了新的原位测量方法,并根据这一测量方法,初步设计制作了相应的海底原位测量仪器。利用初步制作的测量仪器在实验室内对砂质沉积物进行了模拟测量试验,仪器测量得到的沉积物声速与直接透射法测量获得的声速基本一致。在宁波近海海域进行了海上实际测量试验,获得了该测量区域海底底质的声速,验证了此测量技术的可行性,为进一步研发先进的海底原位测量仪器奠定了基础。     

High-Rate Communication for Underwater Acoustic Channels Using Multiple Transmitters and Space–Time Coding: Receiver Structures and Experimental Results

In this paper, we consider the use of multiple antennas and space-time coding for high data rate underwater acoustic (UWA) communications. Recent advances in information theory have shown that significant capacity gains can be achieved by using multiple-input-multiple-output (MIMO) systems and space-time coding techniques for rich scattering environments. This is especially significant for the UWA channel where the usable bandwidth is severely limited due to frequency-dependent attenuation. In this paper, we propose to use space-time coding and iterative decoding techniques to obtain high data rates and reliability over shallow-water, medium-range UWA channels. In particular, we propose to use space-time trellis codes (STTCs), layered space-time codes (LSTCs) and their combinations along with three low-complexity adaptive equalizer structures at the receiver. We consider multiband transmissions where the available bandwidth is divided into several subbands with guard bands in between them. We describe the theoretical basis of the proposed receivers along with a comprehensive set of experimental results obtained by processing data collected from real UWA communications experiments carried out in the Pacific Ocean. We demonstrate that by using space-time coding at the transmitter and sophisticated iterative processing at the receiver, we can obtain data rates and spectral efficiencies that are not possible with single transmitter systems at similar ranges and depths. In particular, we have demonstrated reliable transmission at a data rate of 48 kb/s in 23 kHz of bandwidth, and 12 kb/s in 3 kHz of bandwidth (a spectral efficiency of 4 bs^-1Hz^-1) at a 2-km range.     

CSI feedback-based CS for underwater acoustic adaptive modulation OFDM system with channel prediction

In this paper, we investigate the performance of adaptive modulation （AM） orthogonal frequency division multiplexing （OFDM） system in underwater acoustic （UWA） communications. The aim is to solve the problem of large feedback overhead for channel state information （CSI） in every subcarrier. A novel CSI feedback scheme is proposed based on the theory of compressed sensing （CS）. We propose a feedback from the receiver that only feedback the sparse channel parameters. Additionally, prediction of the channel state is proposed every several symbols to realize the AM in practice. We describe a linear channel prediction algorithm which is used in adaptive transmission. This system has been tested in the real underwater acoustic channel. The linear channel prediction makes the AM transmission techniques more feasible for acoustic channel communications. The simulation and experiment show that significant improvements can be obtained both in bit error rate （BER） and throughput in the AM scheme compared with the fixed Quadrature Phase Shift Keying （QPSK） modulation scheme. Moreover, the performance with standard CS outperforms the Discrete Cosine Transform （DCT） method.