Showing papers on "Adaptive beamformer published in 1999"

A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters

Abstract: This paper proposes a new robust adaptive beamformer applicable to microphone arrays. The proposed beamformer is a generalized sidelobe canceller (GSC) with a new adaptive blocking matrix using coefficient-constrained adaptive filters (CCAFs) and a multiple-input canceller with norm-constrained adaptive filters (NCAFs). The CCAFs minimize leakage of the target-signal into the interference path of the GSC. Each coefficient of the CCAFs is constrained to avoid mistracking. The input signal to all the CCAFs is the output of a fixed beamformer. In the multiple-input canceller, the NCAFs prevent undesirable target-signal cancellation when the target-signal minimization at the blocking matrix is incomplete. The proposed beamformer is shown to be robust to target-direction errors as large as 200 with almost no degradation in interference-reduction performance, and it can be implemented with several microphones. The maximum allowable target-direction error can be specified by the user. Simulated anechoic experiments demonstrate that the proposed beamformer cancels interference by over 30 dB. Simulation with real acoustic data captured in a room with 0.3-s reverberation time shows that the noise is suppressed by 19 dB. In subjective evaluation, the proposed beamformer obtains 3.8 on a five-point mean opinion score scale, which is 1.0 point higher than the conventional robust beamformer.

Theory and application of covariance matrix tapers for robust adaptive beamforming

Abstract: We unify several seemingly disparate approaches to robust adaptive beamforming through the introduction of the concept of a "covariance matrix taper (CMT)". This is accomplished by recognizing that an important class of adapted pattern modification techniques are realized by the application of a conformal matrix "taper" to the original sample covariance matrix. From the Schur product theorem for positive (semi) definite matrices and Kolmogorov's existence theorem, we further establish that CMTs are, in fact, the solution to a minimum variance optimum beamformer associated with an auxiliary stochastic process that is related to the original by a Hadamard (Schur) product. This allows us to gain deeper insight into the design of both existing pattern modification techniques and new CMTs that can, for example, simultaneously address several different design constraints such as pattern distortion due to insufficient sample support and weights mismatch due to nonstationary interference. A new two-dimensional (2-D) CMT for space-time adaptive radar applications designed to provide more robust clutter cancellation is also introduced. Since the CMT approach only involves a single matrix Haddamard product, it is also inherently low complexity. The practical utility of the CMT approach is illustrated through its application to both spatial and spatio-temporal adaptive beamforming examples.

Adaptive filtering of physiological signals using a modeled synthetic reference signal

Abstract: An adaptive filtering method and apparatus for reducing the level of an undesired noise component in an acquired physiological signal having a desired signal component. The acquired physiological signal is applied to one input of the adaptive filter, and a synthetic reference signal that is modeled so as to exhibit a correlation with the desired signal component is applied to another input of the adaptive filter. Thereafter, in a feedback manner, the adaptive filter iteratively adjusts the modeled synthetic reference signal so as to progressively generate a more accurate approximation of the desired signal component in the adaptive filter, which approximation becomes a reconstruction of the acquired physiological signal wherein the level of the undesired noise component is reduced.

Delta-sigma oversampled ultrasound beamformer with dynamic delays

Abstract: The principles of oversampling are exploited in a simple beamforming architecture using a single bit delta-sigma (/spl Delta/C) analog to digital converter (A/D) on every channel. The high sampling rate required for the single bit A/D provides adequate delay accuracy for high quality beamforming using elementary sample manipulations. Images produced with this beamformer exhibit significant artifacts directly related to dynamic focusing. However, a simple digital recording technique following delays permits dynamically focused beamforming without degrading image quality. The simplicity of this beamformer compared to conventional methods may facilitate very large channel count or low power beamformers suitable for 1.5-D arrays or portable scanners.

Interference canceling method and apparatus

Abstract: Interference canceling is provided for canceling, from a target signal generated from a target source, an interference signal generated by an interference source. The beam splitter (114) beam-splits the target signal into a plurality of band-limited target signals band-limited frequency bands and beam-splits the interference signal into corresponding band-limited frequency bands. The adaptive filter (500) adaptively filters each band-limited interference signal from each corresponding band-limited target signal. The beam selector (112) selects beams simultaneously to improve accuracy and, in particular, selects a beam having a fixed direction and a beam which rotates in direction. The noise gate (120) gates the main signal adaptatively filtered by the adaptive filter by opening the noise gate (120) when a signal-to-noise ratio at the near-end is above a predetermined threshold and closing the noise gate when the signal-to-noise ratio at the near-end is below the predetermined threshold. When the target signal represents speech generated at a near end of a teleconference, the adaptive filter (500) cancels an echo present in the reference signal broadcast to a far end of the teleconference.

Software radio architecture with smart antennas: a tutorial on algorithms and complexity

Abstract: There has been considerable interest in using antenna arrays in wireless communication networks to increase the capacity and decrease the cochannel interference. Adaptive beamforming with smart antennas at the receiver increases the carrier-to-interference ratio (CIR) in a wireless link. This paper considers a wireless network with beamforming capabilities at the receiver which allows two or more transmitters to share the same channel to communicate with the base station. The concrete computational complexity and algorithm structure of a base station are considered in terms of a software radio system model, initially with an omnidirectional antenna. The software radio computational model is then expanded to characterize a network with smart antennas. The application of the software radio smart antenna is demonstrated through two examples. First, traffic improvement in a network with a smart antenna is considered, and the implementation of a hand-off algorithm in the software radio is presented. The blocking probabilities of the calls and total carried traffic in the system under different traffic policies are derived. The analytical and numerical results show that adaptive beamforming at the receiver reduces the probability of blocking and forced termination of the calls and increases the total carried traffic in the system. Then, a joint beamforming and power control algorithm is implemented in a software radio smart antenna in a CDMA network. This shows that, by using smart antennas, each user can transmit with much lower power, and therefore the system capacity increases significantly.

Adaptive seismic noise and interference attenuation method

Abstract: A method relating to filtering coherent noise and interference from seismic data by constrained adaptive beamforming is described using a constraint design methodology which allows the imposition of an arbitrary predesigned quiescent response on the beamformer. The method also makes sure that the beamformer response in selected regions of the frequency-wavenumber space is entirely controlled by this quiescent response, hence ensuring signal preservation and robustness to perturbations. Built-in regularization brings an additional degree of robustness. Seismic signals with arbitrary spectral content in the frequency-wavenumber domain are preserved, while coherent noise and interference that is temporally and spatially nonstationary is adaptively filtered. The approach is applicable to attenuation of all types of coherent noise in seismic data including swell-noise, bulge-wave noise, ground-roll, air wave, seismic vessel and rig interference, etc. It is applicable to both linear or areal arrays.

A fast and accurate scheme for calibration of active phased-array antennas

Abstract: The international radio-astronomy community is currently making detailed plans for the development of a new radio telescope: the Square Kilometre Array (SKA). This instrument will be a hundred times more sensitive than telescopes currently in use. One approach for this new telescope is to use a phased array consisting of more than 10/sup 6/ receiving elements with a mixed RF/digital adaptive beamformer. At this moment, a demonstrator receive-only active phased-array system is being tested at NFRA, the One Square Metre Array (OSMA). This is a scale model with 144 receiving elements with a mixed RF/digital beamforming architecture. One of the main issues in getting a high-performance phased-array system with accurate beam control and low side-lobes is an accurate and fast on-line calibration facility. The proposed "multi-element phase-toggle" (MEP) method is an extension of the phase-toggle method used by Lee, Chu and Lin (1993) and allows groups of elements to be calibrated simultaneously by using FFT signal processing techniques. Measured results from the OSMA system performance and of the remaining errors after MEP calibration are presented.

Theoretical assessment of a synthetic aperture beamformer for real-time 3-D imaging

Abstract: A real-time 3-D imaging system requires the development of a beamformer that can generate many beams simultaneously. In this paper, we discuss and evaluate a suitable synthetic aperture beamformer. The proposed beamformer is based on a pipelined network of high speed digital signal processors (DSP). By using simple interpolation-based beamforming, only a few calculations per pixel are required for each channel, and an entire 2-D synthetic aperture image can be formed in the time of one transmit event. The performance of this beamformer was explored using a computer simulation of the radiation pattern. The simulations were done for a full 64-element array and a sparse array with the same receive aperture but only five transmit elements. We assessed the effects of changing the sampling rate and amplitude quantization by comparing the relative levels of secondary lobes in the radiation patterns. The results show that the proposed beamformer produces a radiation pattern equivalent to a conventional beamformer using baseband demodulation, provided that the sampling rate is approximately 10 times the center frequency of the transducer (34% bandwidth pulse). The simulations also show that the sparse array is not significantly more sensitive to delay or amplitude quantization than the full array.

Surrounded by sound

Abstract: The authors discuss immersive audio systems and the signal processing issues that pertain to the acquisition and subsequent rendering of 3D sound fields over loudspeakers. On the acquisition side, recent advances in statistical methods for achieving acoustical arrays in audio applications are reviewed. Classical array signal processing addresses two major aspects of spatial filtering, namely localization of a signal of interest, and adaptation of the spatial response of an array of sensors to achieve steering in a given direction. The achieved spatial focusing in the direction of interest makes array signal processing a necessary component in immersive sound acquisition systems. On the rendering side, 3D audio signal processing methods are described that allow rendering of virtual sources around the listener using only two loudspeakers. Finally, the authors discuss the commercial implications of audio DSP.

Base station transmission beam pattern forming; interference reduction

Abstract: A base station identifies, for a particular or desired mobile user, one or more other mobile users that are to be treated as vulnerable users potentially adversely affected by a downlink signal transmission from the base station to the desired user. A downlink transmission beam pattern is then formed which has a main beam with a direction matching the direction of the desired user and one or more nulls matching the direction or directions of one or more vulnerable users. Transmission data rate to the desired user and/or distance to the desired user may be used as criteria in determining whether this transmission is likely to render other users vulnerable to interference. A user data register (UDR) 14 stores data including direction of arrival (DOA) data for signals received by the base station from the desired and vulnerable users. From data in the UDR 14, a selection unit 16 determines one or more of: the DOA information for the desired user, distances of users from the base station, users who are close to high bit rate users, and users close to clusters of other users. From that data, a list of vulnerable users is generated by the unit 16 which sends the direction information for the desired and vulnerable users to an optimum weights unit 12 which calculates weights input to a downlink beamformer 20 so that the required beam pattern may be formed by transmission antenna elements. Any users for whom provision of a null would degrade the pattern for the desired user are removed from the vulnerable list. Where there is a cluster of vulnerable users, a null may be formed in the mean direction of the cluster. With N transmission antenna elements, N- 1 nulls can be formed, so that if there are more than N-1 vulnerable users, the N- 1 most vulnerable users are selected. The receiving section of the base station may have an adaptive beamformer with an adaptive algorithm 26 which inputs a weight vector to a vector multiplier 22. A direction of arrival (DOA) determining unit 30 estimates the DOA of a user from the weight vector. The unit 30 may estimate the DOA by finding the peak of the uplink beam pattern, eg. by calculating the uplink beamformer gain at different DOAs and identifying the DOA for which the gain is the highest.

Robust space-time adaptive processing (STAP) in non-Gaussian clutter environments

Abstract: The problem of space-time adaptive processing (STAP) in non-Gaussian clutter is addressed. First, it is shown that actual ground clutter returns are heavy-tailed and their statistics can be accurately characterised by means of alpha-stable distributions. Then, a new class of adaptive beamforming techniques is developed, based on fractional lower-order moment theory. The proposed STAP methods adjust the radar array response to a desired signal while discriminating against non-Gaussian heavy-tailed clutter modelled as a stable process. Experimental results with both simulated and actual clutter data show that the new class of STAP algorithms performs better than current gradient descent state-of-the-art methods, in localising a target both in space and Doppler, and thus offers the potential for improved airborne radar performance in STAP applications.

Adaptive multiuser detection and beamforming for interference suppression in CDMA mobile radio systems

Abstract: This paper considers the problem of interference suppression in direct-sequence code-division multiple-access (DS-CDMA) systems over fading channels. An adaptive array receiver is presented which integrates multiuser detection, beamforming, and RAKE reception to mitigate cochannel interference and fading. The adaptive multiuser detector is formulated using a blind constrained energy minimization criterion and adaptation is carried out using a novel algorithm based on set-membership parameter estimation theory. The proposed detector overcomes the shortcomings of conventional LMS- and RLS-type algorithms, namely, that of slow convergence and large computational load, respectively. This is especially the case when strong interferers are present or when the number of adaptive weights is relatively large. DS-CDMA systems can have a relatively large number of spatially distributed interferers. Thus beamforming is based on direction-of-arrival (DOA) estimates provided by an approximate maximum-likelihood estimator (DOA-MLE). Unlike previous approaches, the DOA-MLE exploits the structure of the DS-CDMA signaling scheme resulting in robust performance and simple implementation in the presence of angle spreading. The overall method is suitable for real-time implementation and can substantially improve the interference suppression capabilities of a CDMA system.

A novel wavelet-based generalized sidelobe canceller

Abstract: This paper presents a novel narrow-band adaptive beamformer with the generalized sidelobe canceller (GSC) as the underlying structure. The new beamformer employs a wavelet-based approach for the design of the blocking matrix of the GSC, which is now constituted by a set of regular M-band wavelet filters. Such a construction of the blocking matrix can not only block the desired signals from the lower path as required provided the wavelet filters have sufficiently high regularity, but it also encompasses the widely used one with ones and minus ones along the diagonals as a special case. In addition, it possesses two advantageous features. First, the eigenvalue spreads of the covariance matrices of the blocking matrix outputs, as demonstrated in various scenarios, are decreased as compared with those of previous approaches. Since the popular least mean squares (LMS) algorithm has been notorious for its slow convergence rate, the reduction of the eigenvalue spreads can, in general, accelerate the convergence speed of the succeeding LMS algorithm. Second, the new beamformer belongs to a specific type of partially adaptive beamformers, wherein only a portion of the available degree of freedom is utilized in the adaptive processing. As such, the overall computational complexity is substantially reduced when compared to previous works. The issues of choosing the parameters involved for superior performance are also addressed. Simulation results are furnished as well to justify this new approach.

An efficient scheme for broadband adaptive beamforming

Abstract: This paper introduces an oversampled subband approach to linearly constrained minimum variance adaptive broadband beamforming. This method is motivated by the considerable reduction in computation over fullband implementation and resulting large computational complexity when fullband beamformers with high spatial and spectral resolution are required. We present the proposed subband adaptive beamformer structure, discuss the advantages and limitations of it, and comment on the correct projection of the constraints in the subband domain. In a simulation, the proposed subband structure is compared to a fullband adaptive beamformer, highlighting the benefit of our method.

Adaptive beamforming from the generalized eigenvalue problem with a linear complexity for a wideband CDMA channel

Abstract: We introduce a simple adaptive procedure for computing the suboptimal weight vector of an array system operating at the cell-site. The total computational load for computing the primary eigenvector of the generalized eigen-problem is only O(8.5N) including the matrices updates. The performance of the array system provided by the proposed beamformer is shown in 3 typical cases of multipath fading wideband CDMA channel.

Multichannel adaptive beamforming and interference mitigation in multiuser CDMA systems

Abstract: In CDMA systems, multiple users produce cochannel interference that is only partially mitigated by user-dependent spreading codes. Limitations in any closed-loop transmitter power control can create received power variations at the base station that are potentially large. Multichannel multiuser detectors are data-adaptive receivers that provide interference mitigation as well as spatial and temporal diversity. These receivers are discussed in the context of CDMA applications. A maximum likelihood formulation of multiuser detection with antenna arrays is introduced to provide motivation and intuition. The proposed multiuser receiver incorporates space-time beamforming and time-domain interference cancellation. Performance examples using simulated and experimental data are given along with comparisons with a variety of receiver types.

Adaptive beamforming for an OFDM system

Abstract: This paper presents an adaptive beamforming algorithm for an OFDM system with an adaptive array antenna The proposed algorithm for adaptive beamforming in the OFDM system is derived by (1) calculating the pilot error signals in the frequency-domain, (2) transforming the frequency-domain error signals into time-domain error signals, (3) updating the filter coefficients of the adaptive beamformer in the direction of minimizing the MSE Convergence behavior and performance improvement of the proposed approach are investigated through computer simulation by applying it to the conventional OFDM system

Robust adaptive array beamforming for cyclostationary signals under cycle frequency error

Abstract: This paper deals with the problem of robust adaptive array beamforming for cyclostationary signals. By exploiting the signal cyclostationarity, the SCORE algorithms presented by Agee, Schell and Gardner (1990) have been shown to be effective in performing adaptive beamforming without requiring the direction vector of the desired signal. However, these algorithms suffer from severe performance degradation even if there is a small mismatch in the cycle frequency of the desired signal. In this paper, we first evaluate the performance of the SCORE algorithms in the presence of cycle frequency error (CFE). An analytical formula is derived to show the behavior of the performance degradation due to CFE. Based on the theoretical result, we then develop an efficient method in conjunction with the SCORE algorithms to achieve robust adaptive beamforming against the CFE. Several simulation examples for confirming the theoretical analysis and showing the effectiveness of the proposed method are also presented.

Modal Analysis and Synthesis of Broadband Nearfield Beamforming Arrays

Abstract: This thesis considers the design of a beamformer which can enhance desired signals in an environment consisting of broadband near eld and/or far eld sources. The thesis contains: a formulation of a set of analysis tools which can provide insight into the intrinsic structure of array processing problems; a methodology for near eld beamforming; theory and design of a general broadband beamformer; and a consideration of a coherent near eld broadband adaptive beamforming problem. To a lesser extent, the source localization problem and background noise modeling are also treated. A set of analysis tools called modal analysis techniques which can be used to a solve wider class of array signal processing problems, is rst formulated. The solution to the classical wave equation is studied in detail and exploited in order to develop these techniques. Three novel methods of designing a beamformer having a desired near eld broadband beampattern are presented. The rst method uses the modal analysis techniques to transform the desired near eld beampattern to an equivalent far eld beampattern. A far eld beamformer is then designed for a transformed far eld beampattern which, if achieved, gives the desired near eld pattern exactly. The second method establishes an asymptotic equivalence, up to complex conjugation, of two problems: (i) determining the near eld performance of a far eld beampattern speci cation, and (ii) determining the equivalent far eld beampattern corresponding to a near eld beampattern speci cation. Using this reciprocity relationship a computationally simple near eld beamforming procedure is developed. The third method uses the modal analysis techniques to nd a linear transformation between the array weights required to have the desired beampattern for far eld and near eld, respectively. An e cient parameterization for the general broadband beamforming problem is introduced with a single parameter to focus the beamformer to a desired operating radius and another set of parameters to control the actual broadband beampattern shape. This parameterization is derived using the modal analysis techniques and the concept of the theoretical continuous aperture. A design of an adaptive beamformer to operate in a signal environment consisting of broadband near eld sources, where some of interfering signals may be correlated with desired signal is also considered. Application of modal analysis techniques to noise modeling and broadband coherent source localization conclude the thesis. Glossary of De nitions

Robust adaptive beamforming for wide-band, moving, and coherent jammers via uniform linear arrays

Abstract: The problem of providing robustness to the conventional narrow-band uniform linear array configuration so as to handle wide-band and moving jammers is addressed. This robustness is achieved via the use of derivative constraints in jammer directions. However, since the jammer directions are not known a priori, these constraints are incorporated with a maximum likelihood characterization of the so-called jammer subspace. This formulation does not need to assume the availability of signal-free observations, as stipulated in earlier work. Computer simulation results are presented, which show that the algorithms proposed here yield significantly better performance as compared to the previous algorithms of Gershman et al. (see ibid., vol.44, p.361-6, 1996, and IEEE Trans. Signal Processing, vol.45, p.1878-85, 1997) and Hung and Turner (1983) in a variety of situations required to handle wide-band, moving, and coherent jammers.

Broadband beamforming optimization for speech enhancement in noisy environments

Abstract: We have developed a method to optimize the directional sensitivity of a filter-and-sum beamformer. The directivity of the broadband microphone array is optimized by adjusting the spatial transducer positions and the impulse response of the beamformer. We focus on the optimization of 1-dimensional arrays consisting of M omnidirectional microphones and M FIR filters each of length L. The signal sources are assumed to be point sources evenly distributed over a sphere of radius r or over a set of concentric spheres. The target beam pattern for optimization is defined in terms of the desired source signal directions (mainlobe) and the angles for background noise attenuation.

Signal processing method and apparatus for computing an optimal weight vector of an adaptive antenna array system

Abstract: This invention relates to a signal processing method and apparatus for an adaptive array antenna. The objective is to suggest an adaptive procedure of computing the suboptimal weight vector for an array antenna system that provides a beampattern having its maximum gain along the direction of the mobile target signal source in a blind signal environment, where the transmitted data are not known (or not to be estimated) at the receiver. It is the ultimate goal of this invention to suggest a practical way of enhancing both the communication quality and communication capacity through the optimal weight vector of the array system that maximizes SINR(Signal to Interference+Noise Ratio). In order to achieve this goal, the method of Lagrange multiplier is modified in such a way that the suboptimal weight vector is produced with the computational load of about O(8N), which has been found to be small enough for the real-time processing of signals in most land mobile communications with the digital signal processor (DSP) off the shelf, where N denotes the number of antenna elements of the array.

Generating calibration signals for an adaptive beamformer

Abstract: A beamformer is calibrated for use as an acoustic echo canceler in a hands-free communications environment having a loudspeaker and a plurality of microphones. To perform the calibration, a number of adaptive filters are provided in correspondence with each of the microphones, and each of the adaptive filters is trained to model echo properties of the environment as experienced by the corresponding one of the microphones. A target source is activated, thereby generating an acoustic signal that is received by the microphones. The trained adaptive filters are then used to generate jammer signals by, for example, having each one filter a pseudo noise signal. Respective ones of the jammer signals are then combined with corresponding signals supplied by the microphones, thereby generating combination signals. The combination signals are then used to adapt the beamformer to cancel the jammer signals. In another aspect of the invention, the adaptive filters may be utilized during normal operation by having them perform an echo cancellation operation on each of the signals that is to be supplied to the calibrated beamformer.

Adaptive Beamforming Using a Microphone Array for Hands-Free Telephony

Abstract: This thesis describes the design and implementation of a 4-channel microphone array that is an adaptive beamformer used for hands-free telephony in a noisy environment. The microphone signals are amplified, then sent to an A/D converter. The microprocessor board takes the data from the 4 channels and utilizes digital signal processing to determine the direction-of-arrival of the sources and create an output which ‘steers’ the microphone array to the desired look direction while trying to minimize the energy of interference sources and noise. All of the processing for this thesis will be done on a computer using MATLAB. The MUSIC algorithm is used for direction finding in this thesis. It is shown to be effective in estimating direction-of-arrival for 1 speech source and 2 speech sources that are spaced fairly apart, with significant results down to a -5 dB SNR even. The MUSIC algorithm requires knowledge of the number of sources a priori, requiring an estimator for the number of sources. Though proposed estimators for the number of

Fast adaptive super-exponential multistage beamforming for cellular base-station transceivers with antenna arrays

Abstract: A new blind adaptive beamforming algorithm is introduced. We show how cumulants of the received signals can be used to obtain the weights of the beamformer that perform blind extraction. The method is based on a spatial interpretation of a deconvolution procedure known as the super-exponential algorithm. The basic block processing algorithm is attractive because it can be transformed in an efficient adaptive algorithm which exhibits good tracking capability. To prove the effectiveness of the idea, we show results for a typical mobile communications scenario where several cochannel interferers corrupt the signals of interest.

A low-complexity MVDR beamformer for use with short observation times

Abstract: It is demonstrated, via simulation examples, that the steered, minimum-variance, subarray beamformer is a simple, viable, and computationally efficient approach to partially adaptive beamforming, with the additional advantage that it is available for use with short observation intervals where time is traded for bandwidth. A quadratically constrained version is also examined.

Evaluation of reduced-rank, adaptive matched field processing algorithms for passive sonar detection in a shallow-water environment

Abstract: This paper evaluates the performance of several reduced-rank, adaptive matched field processing (AMFP) algorithms for passive sonar detection in a shallow-water environment. Effective rank reduction improves the stability of adaptive beamformer weight calculation when the number of available snapshots is limited. Here, rank-reduction techniques with various criteria for subspace selection are evaluated within a common framework and compared to the full-rank conventional and minimum-variance (MVDR) beamformers. Results from real data demonstrate that rank reduction, properly applied can improve AMFP detection performance in practical system implementations.

Adaptive beamforming system for radio-frequency interference rejection

Abstract: filter Abstract: Results from an experimental antenna array that is processed adaptively to cancel external radio-frequency interference are presented. The eight-element array has been tested in a far-field anechoic chamber. The adaptive beamformer rejects sidelobe and main-lobe interference by up to 37 dB. A conventional Fourier beamformer rejects interference only in the sidelobes by between 13 and 30 dB depending on the location of the interference within the array's sidelobes. The adaptive beamformer does not need any a priori knowledge about the interfering signals such as the number of interferers or their direction of arrival. The adaptive beamformer also has superior angular resolution to the Fourier beamformer. With an aperture of 4.4 wavelengths and an input signal-to-noise ratio of 37 dB, the adaptive beamformer's resolution is as low as 0.28" while the Fourier beamformer's resolution is 8.8".