IEE Proceedings F (Radar and Signal Processing)
Volume 138, Issue 5, October 1991
Volumes & issues:
Volume 138, Issue 5
October 1991
RCS calculation of open cavities by complex ray expansion
- Author(s): Y.Z. Ruan and W.L. Feng
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 397 –399
- DOI: 10.1049/ip-f-2.1991.0051
- Type: Article
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p.
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A new method of complex ray expansion is employed to calculate the radar crosssection (RCS) for hollow-pipe type targets with large apertures, such as engine intakes. As a typical model, a curved hollow duct is analysed. The effectiveness and accuracy of the method can be verified from the investigation.
Investigation of edge scattering by high-resolution polarisation diversity radar imaging
- Author(s): W.F. Herdeg ; H. Wendel ; U. Fuchs
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 400 –406
- DOI: 10.1049/ip-f-2.1991.0052
- Type: Article
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The paper analyses the polarimetric behaviour of scattering centres showing up in experimental highly resolved two-dimensional radar images of metallic cubes and plates in terms of Kennaugh's and Huynen's target characteristic operator theory. For each scattering centre, the authors determine the dependence of the characteristic angle γ and of the target skip angle η on target aspect θ and frequency. The authors find: (i) cube: GTD simulations accounting only for firstorder wave propagation to and from the cube edges describe the experimental findings; (ii) plates: the frequency dependence of γ and η reflects the interference of the two nonresolved edges of the narrow face of the plate. GTD simulations accounting for higher-order scattering agree reasonable with experiment.
New method of constructing the projection matrix for array processing in single snapshot case
- Author(s): Ming Zhang ; Wanlin Yang ; Lohming Li
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 407 –410
- DOI: 10.1049/ip-f-2.1991.0053
- Type: Article
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p.
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In the paper, a new method of constructing the projection matrix for array processing, using a uniform linear array in the single snapshot case, is presented. The method is computationally simple, independent of the coherence of signal sources and beneficial to real-time processing. Good performance can be obtained when the signal-to-noise ratio (SNR) is high. Simulation results show the effectiveness of the new method.
Waveform analysis for high-frequency FMICW radar
- Author(s): R.H. Khan and D.K. Mitchell
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 411 –419
- DOI: 10.1049/ip-f-2.1991.0054
- Type: Article
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The paper describes the waveform analysis for an experimental long-range highfrequency radar system with a design capability for over-the-horizon detection of ocean surface targets up to 400 km from shore with a range resolution of 400 m. Motivated by the need to maximise the transmitter duty cycle in order to detect long-range targets, the radar is based on a frequency modulated interrupted continuous wave (FMICW) type waveform. The interrupt sequence, which is necessary for a high power monostatic operation, complicates the analysis of the radar performance in terms of the ambiguity function (AF). This paper analyses the range dependent AF of FMICW waveforms and proposes practical interrupt sequences for the synthesis of desired AFs.
Design of noise FM-CW radar and its implementation
- Author(s): Liu Guosui ; Shi Xiangquan ; Lu Jinhui ; Yang Guoyu ; Song Yaoliang
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 420 –426
- DOI: 10.1049/ip-f-2.1991.0055
- Type: Article
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A noise FM-CW radar signal possesses a ‘thumbtack’ average ambiguity function (AAF). The authors present a spectral analysis of the echo signal, and describe two alternative types of signal processing, based on spectral comparison techniques. The two approaches provide a radar that either produces an output when a target is present inside a chosen range, or when it is present between two chosen ranges (giving a zero range ‘hole’ characteristic). The noise FM-CW system has inherently good ECCM properties.
Effectiveness of multinotch logic-product polarisation filters in radar for countering rain clutter
- Author(s): A.J. Poelman and C.J. Hilgers
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 427 –437
- DOI: 10.1049/ip-f-2.1991.0056
- Type: Article
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p.
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The applicability of multinotch logicproduct (MLP) polarisation filters in radar to enhance target detection in a rain environment is investigated. The analyses use backscatter models in terms of polarisation when the environment is illuminated with circular polarisation. These models are, to a great extent, representative of the actual backscatter behaviour of aircraft targets and rain clouds. The degree of polarisation fluctuation of rain cloud returns is a parameter in the analyses. The evaluation results are presented in terms of achievable improvement in the signal-toclutter ratio (SCR) and its dependence on the configuration and number of notches in the MLP filter. As the MLP filter is fundamentally nonlinear, the results provide insight into the upperbound performance of such filters as a function of SCR at the input. Typically, in an environment where a linear single-notch polarisation filter gives a maximum 8 dB improvement in SCR, a nonlinear MLP polarisation filter will give at least 14 dB improvement when the SCR at the input is 0 dB.
Time-domain convolution of one-bit coded radar signals
- Author(s): G. Alberti ; G. Schirinzi ; G. Franceschetti ; V. Pascazio
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 438 –444
- DOI: 10.1049/ip-f-2.1991.0057
- Type: Article
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The compression of the radar signal is considered when the latter and the reference function are both coded at one bit. A sound theory is provided, which describes the performance of hardlimited signals for pulse compression, the effect of noise and the opportunity for fast convolution methods in time domain. The theory is supported by numerical examples on simulated data. Preliminary considerations regarding the use of dedicated hardware are included.
Performance analysis of a noncoherent monopulse processor in the presence of target glint and receiver imbalances
- Author(s): Y. Yoganandam and R. Jitendra Das
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 445 –452
- DOI: 10.1049/ip-f-2.1991.0058
- Type: Article
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Phase imbalances in receiver channels of a monopulse radar can affect its performance adversely. One of the methods suggested to alleviate this problem is the noncoherent processor using linear detection. In the paper, the amplitude based angle (ABA) processor, which uses a squarelaw detector, is considered and explicit expressions are derived for the indicated angle as a function of various imbalances in the receiver channels. Its performance is then compared with that of the exact processor, using shift in null position and deviation in slope of the angle sensitivity curve as criteria. Another serious problem in monopulse processing is the variation of the apparent angular location of targets or glint. In the paper, the ABA processor is analysed and it is shown both analytically and by computer simulations that its performance in the presence of glint is superior to that of the conventional or exact processor.
Constrained adaptation for time delay estimation with multipath propagation
- Author(s): P.C. Ching ; Y.T. Chan ; K.C. Ho
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 453 –458
- DOI: 10.1049/ip-f-2.1991.0059
- Type: Article
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p.
453
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The problem of estimating the time difference between two sensor outputs in the presence of a multipath propagation is addressed. Two adaptive configurations are proposed. One configuration assumes that the multipath delay is an integral multiple of the sampling interval. The other configuration allows a non-integral delay. In either case, filter coefficients are restricted to take on samples of a sine function only. The role of the adaptive filter is to provide appropriate time shifts to the input signals. Incorporating this constraint on the coefficients, which are adapted by the LMS algorithm, reduces the computations and considerably speeds up the convergence rate. Theoretical derivations and computer simulations are presented to demonstrate the effectiveness of the scheme for time delay measurement and to show its ability to accurately track time-varying parameters.
Underwater target tracking via the IRWLS filtering approach
- Author(s): F. El-Hawary and G.A.N. Mbamalu
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 459 –469
- DOI: 10.1049/ip-f-2.1991.0060
- Type: Article
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p.
459
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Underwater target tracking is treated, and a model is reviewed relating the target states to time delay and bearing measurements as the basis of linearised measurement model. The application of conventional filters to the target-tracking problem is reviewed. Problems associated with fitting models using the least squares procedures are addressed. The success of the procedures depends on the assumption that the distribution of the errors resulting from fitting a model to a set of data is Gaussian. For cases of nonGaussian errors, the least squares performance is far from being optimal. Efforts have been made to improve the performance of the least squares procedures for nonGaussian errors, and to enhance their performance for the Gaussian errors. Robust regression procedures appear to perform much better than the least squares procedures when the errors are nonGaussian and also have improved performances for Gaussian errors. Proposed filters based on the iteratively reweighted least squares method are presented, and computational results are offered to illustrate the performance of the techniques. A comparison with Kalman filters, in terms of prediction accuracy and computational time requirements, shows that the proposed filters are advantageous.
Resolution threshold for coherent sources using smoothed eigenstructure methods
- Author(s): L. Chang and C.-C. Yeh
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 470 –478
- DOI: 10.1049/ip-f-2.1991.0061
- Type: Article
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The paper presents a performance analysis of certain eigenstructure based methods (MUSIC type) with suitable forward-only and forward/backward smoothing schemes employed to decorrelated coherent sources. Bias and resolution thresholds are derived for two coherent sources with different phase and unequal power. The effect of phase difference and power ratio on the resolution threshold is discussed. For the case of forward/backward smoothing and two coherent sources with same phase and equal power, the resolution threshold reduces to that obtained by Pillai et al. Detailed computer simulations which confirm the analysis are also presented.
Improvement on the estimation of covariance matrices by incorporating crosscorrelations
- Author(s): R.L. Kirlin and W. Du
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 479 –482
- DOI: 10.1049/ip-f-2.1991.0062
- Type: Article
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p.
479
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The paper addresses the problem of improving the estimate of a covariance matrix from one set of multivariate random processes when there exist nonzero crosscorrelations with another set of random processes. The improvement is obtained by linearly combining the first set's sample covariance matrix with covariance matrices predicted via the crosscorrelations. The superiority of the proposed method is demonstrated by an application to spatial smoothing for the direction-of-arrival (DOA) estimation of coherent narrowband signals using a uniform linear array.
Two-mode adaptive beamformer using a systolic array
- Author(s): K. Takao and C.S. Boon
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 483 –488
- DOI: 10.1049/ip-f-2.1991.0063
- Type: Article
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p.
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An adaptive beamformer using the least squares algorithm usually employs exponential windowing of the input data. In the paper, the authors show that complete suppression of interferences is impossible when an exponential window is applied. To avoid this, a two-mode operation using a systolic array is proposed. Switching is made from an adaptive mode to a frozen mode when the system converges and vice versa when the radio environment changes, by monitoring the change in the magnitude of one of the weights. The proposed method gives exact extraction of the desired signal and is capable of fast suppression of new interferences.
Broadband adaptive beamforming based on coherent signal subspace using spatial interpolation preprocessing
- Author(s): Yuan-Hwang Chen and Feng-Peng Yu
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 489 –494
- DOI: 10.1049/ip-f-2.1991.0064
- Type: Article
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The coherent signal-subspace (CSS) approach to broadband adaptive beamforming for correlated signal and interferences requires preprocessing by transformation matrices to focus signal subspaces at various frequencies into a common one. The transformation matrices often call for preliminary estimates of the directions of arrival of impinging signals through a spatial periodogram, and these preliminary angle estimates, if they are inappropriately selected, especially in the multigroup multiple signals, often result in an estimation error in the transformation matrices, and the array beamforming performance can be then degraded. Therefore an alternative method of preprocessing, called spatial interpolation, which does not require preliminary angle estimates, has been used instead of transformation matrices to estimate the arrival angles of signals impinging on a linear uniform array. The paper extends this idea to broadband adaptive beamforming. Performance comparisons of nulling coherent interferences using CSS with the two sorts of preprocessing are presented. Simulation results show the effectiveness of the spatial interpolation in adaptive beamforming.
DOA estimation with unknown noise fields: a matrix decomposition method
- Author(s): R. Rajagopal and P. Ramakrishna Rao
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 495 –501
- DOI: 10.1049/ip-f-2.1991.0065
- Type: Article
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In this paper, a new method is presented for direction-of-arrival (DOA) estimation in a passive sonar in the presence of unknown correlated noise fields. It is shown that the autocovariance matrix R of received sensor signals can be uniquely decomposed into the sum of two Hermitian matrices. One of these matrices will have column space equal to the signal subspace and the other will have column space orthogonal to the signal subspace. Essential properties of these matrices are identified. These properties are utilised in the matrix decomposition method. Here, the data vector is transformed to another random vector in such a way that the autocovariance matrix R̃ of the transformed vector can be split into the sum of two Hermitian matrices E and F that satisfy the properties identified earlier. It is shown that the noise subspace vectors are then obtained by solving the generalised eigenvalue problem Fx = λR̃x corresponding to λ = 1. Simulation results are also presented to support the theory.
Digital signal processing techniques applied to mobile radio channel sounding
- Author(s): P.C. Fannin ; A. Molina ; S.S. Swords ; P.J. Cullen
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 502 –508
- DOI: 10.1049/ip-f-2.1991.0066
- Type: Article
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A novel implementation of a wideband channel sounder for mobile radio using digital signal processing techniques is presented. The system uses the principle of pulse compression at the receiver. By performing in-phase and quadrature (I and Q) down conversion and matched digital filtering in the frequency domain, an improved dynamic range of 15-20 dB over sounders using SAW matched filters can be achieved. The impulse response of the channel is measured in real-time and thus the sounder does not suffer from any of the compromises associated with the non-real-time sliding correlator sounder. The use of a software approach gives the system great flexibility. In particular, the matched filter compensates for the effect of the amplitude and phase response of the transmitter and receiver filters on the probing pulse.
Noise analysis for digit slicing FFT
- Author(s): Z.A.M. Sharrif ; M. Othman ; T.S. Theong
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 509 –512
- DOI: 10.1049/ip-f-2.1991.0067
- Type: Article
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The digit slicing architecture of the fast Fourier transform is modular in nature and easily applicable to VLSI implementation. The error caused by the digit slicing implementation of on chip fast Fourier transform is analysed. A mathematical model is proposed to arrive at the signal to noise ratio of a general sliced structure by modelling the error as an additive white noise. The model is capable of satisfactorily predicting the signal to noise ratio.
New method of designing adaptive nonlinear filters
- Author(s): K. Deergha Rao and D.C. Reddy
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 513 –519
- DOI: 10.1049/ip-f-2.1991.0068
- Type: Article
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A new method of designing nonlinear filters is proposed. It is shown to be general enough to be applicable to the design of higher order Volterra filters. LMS and fast least squares adaptive algorithms for updating the linear quadratic and cubic weights of a third order nonlinear filter are suggested. Simulation studies are carried out in respect of a third order Volterra filter using the proposed method of design and suggested updating algorithms. The fast least squares implementation is shown to converge to the unknown system parameters considerably faster than the LMS.
LMS adaptive algorithms for joint forward and decision feedback equalisation
- Author(s): D. Ernst
- Source: IEE Proceedings F (Radar and Signal Processing), Volume 138, Issue 5, p. 520 –524
- DOI: 10.1049/ip-f-2.1991.0069
- Type: Article
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The mean-square error is used as the performance measure to find the joint optimum tap settings of an equaliser, consisting of a forward filter and a decision feedback part. This paper provides two LMS adaptive algorithms to attain two different sets of tap settings and shows that one of these sets is suboptimum in the meansquare error sense. In practice the largest possible step size that ensures the algorithm stability must be approximated. Using these step sizes for the iterations, one algorithm converges faster than the other. Switching after initial convergence from the faster algorithm to the slower algorithm yields faster convergence to the minimum mean-square error of the complete equaliser. The results of realised simulations are in good agreement with the calculations.
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