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.     

A low-density extended acoustic sensor for low-frequency arrays

A novel PVDF hydrophone is described which has a sensitivity of -183 dBV/μPa or better and can be made in lengths from 4 cm to more than 1 m, and in diameters from 8 mm to >40 mm. It is based on a piezoelectric PVDF cable which allows considerable design freedom because of its very great mechanical flexibility and high longitudinal piezoelectric coefficient. The design centres on a helical structure which translates a hydrostatic pressure field into a longitudinal compressive strain in the cable and thus makes use of the very sensitive mode. Excellent pressure stability can be achieved, with devices showing a pressure dependency of 0.04 dB/MPa. The device has a flat frequency response from less than 1 Hz to several kHz and a density in the region of 1800 kg/m³, making it suitable for application in low-frequency sonar     

声相关计程仪基阵设计方法

针对声相关计程仪的测速特点,研究了一维、二维基阵的设计方法。一维基阵采用基于约束最小冗余的设计方法,可以获得比均匀线阵大得多的阵列孔径,从而提高基阵的利用率,但约束最小冗余线阵(RMRLA)的设计方法计算量巨大,并不适合二维基阵的设计。在重新定义冗余因子,建立理想位置矢量图模型,提出位置矢量重合率等概念的基础上,实现了适用于声相关测速需求的二维基阵的快速设计。仿真结果证明了该设计方法的可行性。     

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     

Multimegameter-range acoustic data obtained by bottom-mountedhydrophone arrays for measurement of ocean temperature

Acoustic signals transmitted from the ATOC source on Pioneer Seamount off the coast of California have been received at various sites around the Pacific Basin since January 1996. We describe data obtained using bottom-mounted receivers, including US Navy Sound Surveillance System arrays, at ranges up to 5 Mm from the Pioneer Seamount source. Stable identifiable ray arrivals are observed in several cases, but some receiving arrays are not well suited to detecting the direct ray arrivals. At 5-Mm range, travel-time variations at tidal frequencies (about 50 ms peak to peak) agree well with predicted values, providing verification of the acoustic measurements as well as the tidal model. On the longest and northernmost acoustic paths, the time series of resolved ray travel times show an annual cycle peak-to-peak variation of about 1 s and other fluctuations caused by natural oceanic variability. An annual cycle is not evident in travel times from shorter acoustic paths in the eastern Pacific, though only one realization of the annual cycle is available. The low-pass-filtered travel times are estimated to an accuracy of about 10 ms. This travel-time uncertainty corresponds to errors in range- and depth-averaged temperature of only a few millidegrees, while the annual peak-to-peak variation in temperature averaged horizontally over the acoustic path and vertically over the upper 1 km of ocean is up to 0.5°C     

Flattening the side-lobes of wide-band beam patterns [acoustic arrays]

Recently, simulated annealing has been proposed as a suitable method for optimizing the weighting window of a linear array working under wide-band conditions. This paper introduces and discusses a new energy function that, minimized by simulated annealing, produces a beam pattern showing a flat side-lobe profile over a large interval.     

Programmable switched-capacitor neural network for MVDR beamforming

In this paper, a real-time adaptive antenna array based on a neural network approach is presented. Since an array operating in a nonstationary environment requires a programmable synaptic weight matrix for the neural network, the switched-capacitor (SC) circuits with the capability of programmability and reconfigurability is conducted to implement the neural-based adaptive array. Moreover, the SC techniques can directly implement the neural network with less chip area and provide the ratio of SC-equivalent resistors with accuracy of 0.1 percent. Programming of the switched-capacitor values could be made by allocating each synaptic weight to a set of parallel capacitors with values in a digitally programmable capacitor array (PCA). A relatively wide range of values (5 to 10 binary bits resolution) can be realized for each synaptic weight. A simulation tool called SWITCAP is used to verify the validity and performance of the proposed implementation. Experimental results show that the computation time of solving a linear array of 5 elements is about 0.1 ns for 1 ns time constant and is independent of signal power levels     

An efficient architecture for real-time narrowband beamforming

A short finite impulse response (FIR) filter architecture for narrowband beamforming is presented that is well suited for active sonar applications. By applying the technique of adaptive modeling, a short FIR filter can be designed to carry out any required narrowband constant phase shift. A 4-tap FIR filter designed in this way can generate constant phase shift within bandwidths of 0.04 (relative to the sampling frequency), with an error of less than 1°. Thus a beam can be formed with a bank of short FIR filters, each filter corresponding to one sensor. Due to the light computational load of this method, it is rather convenient for digital signal processors (DSP) to implement beamforming in real time. Satisfactory sonar beam patterns are shown to result from a TMS320C25-based emulation of the architecture     

True time-delay bandpass beamforming

A simple digital scheme for bandpass time-domain beamforming that is applicable when there is significant change in the envelope of the signal across the extent of an array of hydrophones is presented. The three requisite processes to obtain complex-baseband beam signals, namely, down conversion to complex baseband, time delay of the complex envelope, and phase rotation, are discussed. Down conversion is accomplished by subsampling the bandpass signal and an efficient interpolation technique is employed for time registration of these samples. Since the same interpolation technique is u sed to implement the time delay, the two processes are combined. With further modification to effect the phase rotation, it is shown how the three requisite processes for bandpass beamforming can be accomplished with a single analog-to-digital converter and two short finite impulse response filters     

The TAP III beamforming system

The Three-Array Processor (TAP III) beamforming system incorporating both wide-band time-domain beamforming and narrow-band frequency-domain beamforming is described. This paper briefly develops the beamforming theory and shows how the fast Fourier transform (FFT) is applied to accomplish frequency-domain beamforming. The frequency-domain beamformer operates in the frequency domain to form beams and power spectrum data over narrow frequency bands of interest. A real-time digital filtering technique is used to extract the narrow bands of interest from the broad-band input signal. The frequency-domain beamformer accomplishes real-time digital filtering and beamforming by using a high-speed array processor to do the complex calculations and data handling required by the algorithm. The time-domain beamformer operates in parallel with the frequency-domain beamformer to form up to 16 broad-band beams in the time domain. A programmable all-pass digital filter is used to create the fine time delays required by the time-domain beamformer.     

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     

A methodology for acoustic seafloor classification

A seafloor classification methodology, based on a parameterization of the reverberation probability density function in conjunction with neural network classifiers, is evaluated through computer simulations. Different seafloor provides are represented by a number of scatterer distributions exhibiting various degrees of departure from the nominal Poisson distribution. Using a computer simulation program, these distributions were insonified at different spatial scales by varying the transmitted pulse length. The statistical signature obtained consists of reverberation kurtosis estimates as a function of pulse length. Two neural network classifiers are presented with the task of discriminating among the various scatterer distributions based on obtained acoustic signatures. The results indicate that this approach offers considerable promise for practical, realizable solutions to the problem of remote seafloor classification     

Acoustic telemetry for ocean acoustic tomography

Very low data rate burst telemetry for long-range deep-ocean applications is discussed. Energy efficiency, propagation coherence, and waveform coding, together with transmitter constraints, influenced the design of proposed buoy-to-buoy and buoy-to-shore systems.     

Adaptive beamforming: spatial filter designed blocking matrix

Controlling the resolution in adaptive beamformers is often crucial. A simple method that works for both narrow-band and broad-band arrays is presented. This method is based on the normalized leaky LMS algorithm in conjunction with a generalized sidelobe canceller (GSC) structure, where the GSC is designed using a spatial filtering approach. In essence, the suppression of the spatial filters and the implicit noise of the leaky LMS algorithm together determine the adaptive beamformer. Analytical expressions are given for the Wiener filters and the output spectrum versus frequency and point source location. These expressions are employed in the design specification of the spatial filters and to obtain conditions for a controlled quiescent beamformer response. Simulation results are presented to illustrate the behavior of the array     

Code-fed omnidirectional arrays

The synthesis of linear and planar arrays of desired omnidirectional intensity patterns is considered. A novel approach that utilizes the relation between the array intensity pattern and the correlation function of the used feeding sequence, or code, is adopted. The basic theory of such code-fed arrays is reviewed, and it is shown that almost omnidirectional patterns result when codes with sharp autocorrelation functions are used as the feeding sequences. Examples of omnidirectional linear and planar arrays fed with Barker codes, Kuttruff-Quadt trial-and-error two-dimensional binary codes, and nonbinary Huffman-type codes are presented. The results have direct application in underwater communication systems, public address systems, and acoustical imaging systems     

Digital waveform codings for ocean acoustic telemetry

M-sequence waveform coding with a single long codeword has been considered as the basis for long-range underwater acoustic telemetry for one user. (An m-sequence is a periodic, binary, linear-law maximal-length sequence. If the span of the law is n, the maximal length L-2ⁿ=1). For a given law, a single m-sequence transmits a maximum of log₂ (L) bits of source information per channel word. To increase the number of bits per word, families of m-sequences and Gold codes are considered and compared to a single m-sequence. A hypothetical idealized multipath channel with added white Gaussian noise is assumed. Coding using families of m-sequences is recommended because it requires a smaller bit-energy-to-noise ratio than other waveform codes to achieve an equivalent codeword error probability     

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     

Adaptive equalization techniques for acoustic telemetry channels

A tutorial review of adaptive equalization techniques for combating intersymbol interference in high-speed digital communications over time-dispersive channels is given. Various equalizer structures and the associated adaptive algorithms, including both fractionally spaced and symbol-spaced equalizers, are presented. Also considered is the application of adaptive equalization techniques to underwater acoustic telemetry channels