Showing papers on "Adaptive beamformer published in 1980"
09 Apr 1980
TL;DR: An analysis of this technique is extended to the case when a linear filter appears in the auxiliary signal path and a general solution to this problem is obtained.
Abstract: A technique known as a "multiple correlation cancellation loop" and also as the "LMS algorithm" is widely used in adaptive arrays for radar, sonar, and communications, as well as in many other signal processing applications. In this paper, an analysis of this technique is extended to the case when a linear filter appears in the auxiliary signal path. A general solution to this problem is obtained and several examples for narrowband and broad-band signals are presented.
219 citations
TL;DR: Certain properties of the eigenvectors of the correlation matrix of an array illuminated by a field of correlated discrete sources are established and an adaptive algorithm is proposed for resolving the direction of all discrete sources, even if they are mutually correlated.
Abstract: Adaptive beamforming algorithms are often designed under the assumption that the sources illuminating the array are mutually uncorrelated. When the assumption is not valid, the performance of the adaptive beamforming algorithm in certain applications is severly limited. A simple example is included which illustrates how suppression of a look direction signal occurs when a power minimizing adaptive algorithm is used in the presence of correlated sources. This paper establishes certain properties of the eigenvectors of the correlation matrix of an array illuminated by a field of correlated discrete sources. On the basis of these properties an adaptive algorithm is proposed for resolving the direction of all discrete sources, even if they are mutually correlated. Simulation results are presented showing the performance of the algorithm proposed in this paper.
68 citations
TL;DR: It is demonstrated that for time multiplex perturbation sequences the penality incurred because of the use of perturbations for gradient estimation in adpative beamforming is an increase by a factor N in the system time constants.
Abstract: The application of orthogonal perturbation sequences to adaptive beamforming is considered. The perturbation approach is relevant when all the array element signals in an array are not accessible but the array weights are independently adjustable. The perturbation sequences can be used to provide an estimate of the gradient required for adaptive beamforming. A unified analysis of the performance of orthogonal perturbation sequences is presented, and the performance of some specific sequences is investigated. It is demonstrated that for time multiplex perturbation sequences the penality incurred because of the use of perturbations for gradient estimation in adpative beamforming is an increase by a factor N in the system time constants.
56 citations
TL;DR: The problem of estimating the strengths of signals arriving at an array of receivers when the arrival directions are known, is addressed and several new estimators are derived and relationships between them and the conventional and adaptive beamformers are given.
Abstract: The problem of estimating the strengths of signals arriving at an array of receivers when the arrival directions are known, is addressed. Several new estimators are derived and relationships between them and the conventional and adaptive beamformers are given. Algorithms are provided for numerical solutions for these estimators, with examples using experimental data. The techniques are not confined to sensor-array processing but can be applied to a wide range of other estimation problems.
28 citations
TL;DR: A novel two-stage adaptive signal extractor for intermittent signal applications that will adapt only when the signal is present and thereby effect a reduction in the distortion caused by the gust stage is presented.
Abstract: A novel two-stage adaptive signal extractor for intermittent signal applications is presented. If the presence and absence of the signal can be detected, the first stage will adapt only while the signal is absent and thereby effect a reduction in noise, whereas the second stage will adapt only when the signal is present and thereby effect a reduction in the distortion caused by the gust stage. Bounds on performance are derived, and performance improvement relative to a conventional one-stage adaptive noise canceller is assessed.
27 citations
01 Dec 1980
TL;DR: An approach to the adaptive realization of the multipath focused beamformer is given which exploits an orthogonal decomposition of the estimated spatial cross-spectral density matrix (CSDM) of the array.
Abstract: The maximum likelihood procedure for estimation of source location parameters in a multipath medium using a passive array of point sensors is presented. The maximum likelihood estimation procedure is explained in terms of combined focused beamformer operations on each multipath arrival. An approach to the adaptive realization of the multipath focused beamformer is given which exploits an orthogonal decomposition of the estimated spatial cross-spectral density matrix (CSDM) of the array. An example from passive sonar array processing is given.
5 citations
TL;DR: In this paper, a comprehensive model study on the use of horizontal hydrophone arrays in shallow water is presented, where well-known signal processing techniques are applied to a shallow-water sound-propagation model in order to investigate the particular influence of shallowwater conditions on the design of spatial receiver structures.
5 citations
TL;DR: It is shown that by choosing appropriately between minimization and maximization, a useful desired signal-to-interference-plus-noise ratio can be maintained at the array output over a wide range of signal powers.
Abstract: A weight control algorithm that may be used in a two-element adaptive array to protect a desired signal from an interference signal is described. The algorithm is a constrained gradient technique that either maximizes or minimizes array output power. It is shown that by choosing appropriately between minimization and maximization, a useful desired signal-to-interference-plus-noise ratio can be maintained at the array output over a wide range of signal powers.
5 citations
01 Sep 1980
TL;DR: In this paper, various methods of frequency-domain beamforming for a line array of hydrophones were examined, including the conventional phase-to-plane beamformer, the data-adaptive minimum variance beamformer and the principal solution beamformer.
Abstract: : The paper considers various methods of frequency-domain beamforming for a line array of hydrophones The line array geometries are restricted to uniform sensor spacings (ie filled array) or a geometry in which sensors are eliminated from the filled structure (sparse array) The methods to be examined include the conventional phase-to-plane beamformer (CB), the data-adaptive Minimum Variance beamformer (MVB), and the Principal-Solution beamformer (PSB) The beamformer performance is assessed on the basis of two criteria The first is the ability of the beamformer to map the directionality of the acoustic field The integration time uesd in producing the map is unrestricted: hence, only first-order beamformer statistics influence the performance The second criterion is the ability to detect narrowband, directional signals against the noise Here the integration time is finite, and higher-order beamformer statistics become important We shall show that the PSB has advantages over the CB for mapping of the directionality However, no advantage exists in the signal detection application The MVB provides the directionality map that is least influenced by leakage However, its detection performance is superior to that of the CB only if the noise field is highly directional (Author)
3 citations
TL;DR: In this article, an existing adaptive beamforming technique, known as frequency-domain adaptive interference rejection (FAIR), is examined for robustness with respect to nonsystematic array errors.
Abstract: An existing adaptive beamforming technique, known as frequency‐domain adaptive interference rejection (FAIR), is examined for robustness with respect to nonsystematic array errors. It is concluded that, for array position errors below ±10% of the mean spacing, the FAIR technique continues to be effective. However, for special situations where large errors are encountered or where the noise situation is so favorable that better interference rejection is worthwhile, error compensation might become necessary, and a self‐calibration scheme is briefly described.
2 citations
09 Apr 1980
TL;DR: Design trade-offs in a programmable time-delay-and-sum digital beamformer built by MPL permits the incorporation of slow changes in element positions and/or beam steering direction while the beamformer carries out the real-time formation of 1300 beams from 32 input sensors.
Abstract: Design trade-offs in a programmable time-delay-and-sum digital beamformer built by MPL will be discussed. This dynamic beam-former permits the incorporation of slow changes in element positions and/or beam steering direction while the beamformer carries out the real-time formation of 1300 beams from 32 input sensors. The sensors are distributed in a random but known manner over an aperture diameter of up to 800 λ (at 400 Hz).
TL;DR: A method for achieving an optimal allocation of fixed computational resources in a spatial/spectral filtering system is developed and two simulation examples are presented to indicate the kind of improvement that is possible in a system design.
Abstract: A method for achieving an optimal allocation of fixed computational resources in a spatial/spectral filtering system is developed. Specifically, clock cycles per unit time for a given processor are to be optimally allocated. The system is a cascade of three subsystems: an adaptive beamformer, a bandpass filter, and a spectral whitener. Each subsystem has adjustable parameters which are related to required arithmetic operations. A gradient projection algorithm is used to find a best allocation according to specified criteria. Two simulation examples are presented to show how the algorithm operates and to indicate the kind of improvement that is possible in a system design.
TL;DR: The mathematical operations performed in a phase shift beamformer are recorded in a more efficient manner, and the result is an implementation that minimizes the hardware needed.
Abstract: The mathematical operations performed in a phase shift beamformer are recorded in a more efficient manner. The result is an implementation that minimizes the hardware needed. The performance of an experimental beamformer using this architecture is assessed; the beam pattern of the phase shift beamformer compares favorably to that of a time delay beamformer. Finally, the implications of integrating the architecture in a single integrated circuit are investigated.