Showing papers on "Beamforming published in 1977"

Beamformer for wideband signals

Abstract: A beamformer for wideband signals wherein the elemental signals available the sensors of a receiving array are subjected to Fast Fourier Transformations which decompose them into a plurality of narrowband signals. The narrowband signals, which consist of signals having the same Fourier coefficients, are subjected to appropriate phase shifts to form a plurality of narrowband beams having the directional characteristics desired. Reconstruction of the wideband signal is accomplished by subjecting the narrowband beams to an inverse Fast Fourier Transform. The beamforming is thus performed with narrowband signals, and only phase shift circuits are required.

Frequency domain beamforming techniques

Abstract: It is well known that sensor signals must be sampled at a rate much higher than the Nyquist rate to minimize degradations of side lobe levels and maximum response axis gain in sampled time domain beamformers. Various frequency domain techniques can be employed to form beamswithout requiring this excessively high data rate that can cause data management problems in programmable signal processors. Techniques applicable to line, ring, cylindrical and spherical arrays are developed.

Linear Signal Processing Architectures

Abstract: A large portion of the computational load for many signal processing problems consists of the computation of linear transforms. For time-invariant linear transforms such as cross convolution or matched filtering, the transversal filter provides a highly parallel computational module with high throughput and minimal control overhead. This paper will show how similar computational modules can be configured to provide similar computational advantages for a large class of timevariant linear transforms including one-dimensional and multi-dimensional discrete Fourier transforms and one-dimensional and two-dimensional discrete cosine transforms. Furthermore, time-variant transform modules may be combined to implement high capacity time-invariant linear transforms. The implementation of these techniques using surface acoustic wave (SAW) and charge coupled device (CCD) technology permits the real-time solution of several important signal processing problems including image data compression, spectrum analysis, convolutional array scanning and beamforming. Advanced digital and integrated analog/digital architectures will permit these fast processing techniques to be extended to high accuracy and adaptive processing tasks.

Phased array antenna control

Abstract: The present invention provides several new and useful improvements in steering and control of phased array antennas having a small number of elements, typically on the order of 5 to 17 elements. Among the improvements are increasing the number of beam steering positions, reducing the possibility of phase transients in signals received or transmitted with the antennas, and increasing control and testing capacity with respect to the antennas.

Maximum‐likelihood estimation of source parameters from time‐sampled outputs of a linear array

Abstract: This paper considers the joint estimation of the bearing and strength parameters of a noise source, by a uniformly spaced array of sensors in the presence of self‐noise, assumed to be independent between any pair of sensors. Contrary to the analogue systems of processing (i.e., correlation and beamforming), the present scheme, which is based on the maximum‐likelihood (ML) principle, operates on a set of time samples representing the bandlimited output of the array’s elements. The system, as a result, does not require the assumption of long observation time normally used in other schemes and is easily implemented on a digital computer. The resulting ML estimator is not in theory a sufficient one. Nevertheless, when the estimator’s variance is compared with the Cramer–Rao lower bound, the estimator virtually attains its asymptotic sufficiency as the number of array elements exceeds a ’’threshold’’ which is a decreasing function of the signal‐to‐noise ratio (SNR) and the length of the observation time. Except when the SNR is very poor and the observation time is quite short, the ’’threshold’’ is found to be surprisingly small. It is also demonstrated, at least when the error in the bearing estimate is small, that the ML estimator is unbiased.

Digital beamsteering for a parametric scanning sonar system

Abstract: Digital beamsteering for a parametric scanning sonar steers a nonlinear, ulated signal of a single high frequency by appropriately delaying and/or varying the phase of the signal as it enters each of a plurality of elements in an array. The sonar beam can be steered electronically over a wide range of angles.

Sonar imaging with the synthetic aperture technique

Abstract: It is well known that the coherency content of acoustic signals propagated in the ocean has been exploited in signal processing schemes to enhance the signal-to-noise performance of sonar systems. Currently, acoustic imaging techniques which rely on coherency are being developed. Recently, some of these techniques have passed from laboratory curiosities to hardware systems. Two such applications are acoustic holography and synthetic aperture sonar (SAS). This paper reviews the basic principles of the SAS technique. The characteristics which are useful for some present day problems will be discussed. Limitations of the SAS method caused by the ocean environment will be noted in general terms.

Solving the sonar equations with fluctuating signals in noise

Abstract: Fluctuations of signals and noise are commonplace in sonar and have profound effects on the detectability of sonar targets. In this paper, "transition curves", giving probability of detection vs "signal excess", are derived for three theoretical fluctuation models and their use in making sonar performance predictions by means of the sonar equations is described. Some detection results from fleet exercises are presented. They seem to show that the handy and oft-assumed log-normal fluctuation model, with a σ of 6 to 8 dB, provides as good a fit as any to the sparse and scattered field data.

Two-dimensional transform processor for sonar beamforming using c.c.d. and s.a.w devices

Abstract: A 2-dimensional transform technique which can simultaneously process both target Doppler and target bearing information for sonar beamforming applications is reported. The system uses charge-coupled-device (c.c.d.) analogue time compression on each hyprophone element, permitting temporal transformation of the data to be performed in a surface-acoustic-wave (s.a.w.) complex-Fourier-transform processor. Transformed data are selected from successive hydrophone records by Doppler cell, and a second transform, effectively a spatial transform, is performed on each Doppler cell. The resultant target bearing and target Doppler information is displayed as a B-scan.

Adaptive cross‐sensor beam forming with planar arrays

Abstract: In cross‐sensor beamforming the narrow‐band quadrature signal at one sensor is multiplied by the complex conjugate of the narrow‐band quadrature signal at a second sensor to form a stable element of the cross‐sensor field. After suitable time averaging, the beam response can be calculated as the weighted correlation between the cross‐sensor field that is measured and the field that is expected. If N sensors form a sparse line array, then N lines form a sparse square array. Examples are given showing the performance of a sparse square array and a random planar array in resolving a realistic multiwave signal plus noise field. A simple adaptive algorithm is used to resolve weak signals close to strong signals and to reduce sidelobe problems associated with a random array.

Array Design for Measuring Source Depth in a Horizontal Waveguide

Abstract: The signal-to-noise ratio of received radiation is enhanced by properly combining the outputs from a collection of sensors which are connected as an array. In an infinite homogeneous medium, such as a deep ocean layer, the array is matched to the incoming signal by adjusting the phases or time delays of the sensor outputs so that the signals may be added coherently. Beamforming, however, is inappropriate if the waveguide boundaries significantly effect the radiation field. The maximum information array design for estimating source depth in an infinite homogeneous horizontal waveguide is shown to be horizontal, and thus a horizontal array can be used to obtain source depth as well as bearing in a shallow waveguide.

Electronically Scanned and Focused Receiving Array

Abstract: A ninety element electronically scanned and focused receiving array has been built and tested. The discrete transducer elements are formed by partial dicing of PZT-5 ceramic bars. Focused beamforming is accomplished by a hybrid analog-digital electronic system. Correct phasing of the transducer elements is achieved by multiplying the received 500 kHz acoustic signals by a low frequency FM sweep or “chirp” signal. The chirp, whose frequency range determines the focal distance, is generated by a VCO, digitized, and entered into a four-level digital delay line ninety units in length. An output tap on the delay line is provided for each hydrophone. Multiplication of the chirp and the acoustic carrier is accomplished by an integrated circuit digital-to-analog multiplier. The current outputs of the multipliers are added by a network of summing amplifiers. After high pass filtering, an amplitude modulated 500 kHz carrier remains. Single sideband demodulation is used to produce the scanning beam output. This signal is then rectified and low-pass filtered and can be used to intensity modulate a CRT display. The system can focus down to a range of less than one meter and produces a resolution of the order of one centimeter at that distance.

Digital processing of Doppler sonar signals

Abstract: The signal processing gain by digital beamforming in Doppler sonar was investigated in terms of the enhancement of signal‐to‐reverberation ratio (SRR) at the postbeamforming Doppler‐filter output with an objective to determine the minimum number of bits necessary for the quantization of beamformer channel input signals without appreciable degradation of sonar performance. The formulation of the problem follows closely the author’s earlier work which extends Davenport’s analysis of bandpass limiter to the more general multichannel multibit signal processors. Owing to the Doppler shift of target signal frequency and the relatively narrow bandwidth of Doppler filters, we express the frequency selective effect in the nonlinear interacting power series in terms of a new set of coefficients cmk, which bear crucial significance in sonar performance. Extensive numerical calculations of ’’Doppler processing gain’’ were carried out to demonstrate that four‐bit quantization of array transducer signals is theoretically sufficient to yield a performance essentially identical to that of a linear beamformer, but severe loss of gain could result if fewer than four bits are taken at the quantizer output. It is further demonstrated numerically and substantiated analytically that, for small input SRR, the processing gain of a digital beamformer with Doppler filters cannot exceed that of a linear (or optimized four‐bit digital) beamformer even it its channel signals are quantized into more than four bits.

The Effects of Clipping on Linear Beamforming With Sinusoidal Inputs.

Abstract: : The problem addressed in this thesis is the overload of a linear transducer array. Under normal conditions, such overload results in clipping of the input signal. Such clipping may occur at the array elements or at the output of the beamformer with differing results. This paper presents a comparative study of the effect on beamformer performance of clipping at the transducer elements and of clipping at the beamformer output for sinusoidal signal and interference. The study was made using a computer simulation of a linear, single-line array connected to a bandpass spectrum analyzer. It was found that even hard clipping, i.e., clipping at a level far below that of the input signals, at the transducer elements had very little effect on the directivity of the array. Clipping at the output of the beamformer was found to have considerable effect on the directivity. This effect was strongly dependent upon the signal-to-interference ratio and upon the signal level at which clipping occurred. It was concluded that the beamformer should be designed so as to guarantee that if clipping is to occur, it be made to take place at the elements. (Author)

High resolution sonar simulation techniques

Abstract: A mathematical model developed for a computer simulation of both side-looking and forward-looking high resolution sonar systems is described. The purpose of the simulation is to permit systematic examination of the effects of first-order parameter variations on sonar image quality and target detection capability. The model permits specification of the major parameters such as height above bottom, bottom reflectivity, projector power and directivity function, receiver array geometry, number of receiver elements, shading, detection bandwidth, and display characteristics. Details of the model and representative simulated displays for various sonar parameter combinations are presented.

Digital interpolation beamforming; concept and theory

Abstract: For many applications, digital beamformers require an input rate which is significantly higher than that required for waveform reconstruction to achieve an adequate set of synchronous beams. Typically, this high rate is realized in the A/D conversion of the sensor outputs. A novel alternative, discussed in this paper, is to sample the sensors at a rate consistent with the Nyquist criterion and implement vernier beamformer delays by digital interpolation. This technique, which is referred to as digital interpolation beamforming, permits an efficient partitioning between A/D converter and cable bandwidth requirements and digital processing complexity. The basic structure of an interpolation beamformer is presented. It is shown that interpolation filtering and beamforming can be interchanged to minimize digital processing. Beam‐pattern degradation due to interpolation error is derived and interpreted for the special case of a line array.

An Experimental Digital Beamforming Sonar

Abstract: Previous experimental results using a 250-kHz acoustic imaging system show the Potential usefulness of holographic processing in high resolution sonars 1,2 This paper presents the design and preliminary experimental results of a holographic sonar using a linear array, As presently configured, the system consists of a 128-element line array, a single preamplifier/sine-cosine channel, a cylindrical acoustic projector, an Altair 8800 digital microcomputer, and an HP 1335A storage display.

Fast Beamforming Processor

Abstract: A beam former implementation is described which combines the computational efficiency of a Fast Fourier Transform algorithm with the speed and economy of analog signal processing hardware. The fast transform algorithm enables a single processor module to provide 32 simultaneous beams when used with a line array of 32 equidistantly-spaced transducers. The signal processing required to implement this function is the equivalent of performing, in real-time, a 32 point Fast Fourier Transform on 32 continuous 100 kilohertz input signals, 100 kilohertz being the isonifier frequency. The fast analog transform technique allows this amount of processing to be performed on a single circuit board, whereas a digital implementation would require a great number of high speed calculations. The operational amplifier circuit configurations which perform the multiplications and summations, and the algorithm characteristics which facilitate analog implementation, are described. The use of multiple Fast Beamforming Processor modules for the real-time generation of two-dimensional ime.es is also discussed.

Adapt i ve array process i ng experi ments at hf1

Abstract: This paper summarizes experimental results which demon­ strate that significant improvements in signal-to-noise ratio SNR can be achieved by utilizing adaptive array methods in a 15 MHz, electromagnetic, high frequency (HF), radar system. The experi­ ments were conducted with a bistatic radar which employed adaptive beamforming at the receive array elements only. Two adaptive algorithms were studied and in both cases it was shown that signal­ to-noise ratio improvements of 10 to 15 dB are readily achieved when adaptive beamforming is compared with conventional, Dolph­ taper beamforming methods using identical received data in an HF backscatter environment. It was also demonstrated that the time scale of coefficient variation in an adaptive processor operating in this environment is the order of one second. Successful track­ ing of the adaptive algorithm under these conditions was demon­ strated. The use of MTI clutter suppression filters at the sub­ array outputs, prior to adaptation, was investigated. No signifi­ cant improvement was observed with the use of these filters on experimental data. Finally, it was shown that the presence of fading nulls can significantly affect the determination of optimal subarray location and spacing in an HF environment. In general, the adaptive beamformer performance was found to be less dependent upon array geometry than was the case for conventional processing.

Solving signal processing algorithms with a multi-microprocessor network

Abstract: This paper presents a state-of-the-art approach to a classic sonar signal processing problem. First a theoretical foundation for Source Oriented Beam Forming (SOBF) is introduced. A distributed processing element is then introduced and discussed which is offered as one solution to the platform implementation of high-powered signal processing such as SOBF. While many of the details of the implementation are beyond the scope or intent of this paper they may be found in the references.

Adaptive Array Processing Experiments at HF

Abstract: This paper summarizes experimental results which demonstrate that significant improvements in signal-to-noise ratio SNR can be achieved by utilizing adaptive array methods in a 15 MHz, electromagnetic, high frequency (HF), radar system. The experiments were conducted with a bistatic radar which employed adaptive beamforming at the receive array elements only. Two adaptive algorithms were studied and in both cases it was shown that signal-to-noise ratio improvements of 10 to 15 dB are readily achieved when adaptive beamforming is compared with conventional, Dolphaper beamforming methods using identical received data in an HF backscatter environment. It was also demonstrated that the time scale of coefficient variation in an adaptive processor operating in this environment is the order of one second. Successful tracking of the adaptive algorithm under these conditions was demonstrated. The use of MTI clutter suppression filters at the subarray outputs, prior to adaptation, was investigated. No significant improvement was observed with the use of these filters on experimental data. Finally, it was shown that the presence of fading nulls can significantly affect the determination of optimal subarray location and spacing in an HF environment. In general, the adaptive beamformer performance was found to be less dependent upon array geometry than was the case for conventional processing.