Array Signal Processing

This paper describes the use of a 550 (25x22) element 2MHz 2D piezoelectric composite array in immersion mode to image an aluminum test block containing a collection of artificial defects. The defects included a 1mm diameter side-drilled hole, a collection of 1mm slot defects with varying degrees of skew to the normal and a flat bottomed hole. The data collection was carried out using the full matrix capture; a scanning procedure was developed to allow the operation of the large element count array through a conventional 64-channel phased array controller. A 3D TFM algorithm capable of imaging in a dual media environment was implemented in MATLAB for the offline processing the raw scan data. This algorithm facilitates the creation of 3D images of defects while accounting for refraction effects at material boundaries. In each of the test samples interrogated the defects, and their spatial position, are readily identified using TFM. Defect directional information has been characterized using VTFM for defect exhibiting angles up to and including 45 o of skew.

Özetçe—GPS alıcılarının işlevleri istemli/istemsiz karıştırıcılar tarafından düşük güçlerle bile engellenebilmektedir. Karıştırıcılar yaygınlaştıkça görev kritik alıcıların karıştırmaya karşı korunması önem kazanmaktadır. Bu çalışmada anten dizisi yardımıyla uzaysal filtreleme yapılarak GPS L1 bandında karıştırıcı bastırımı hedeflenmiştir. Yapılan ön çalışmaların ardından gerçek ortam denemelerinde karıştırmaya maruz kalan GPS alıcısının çalışmaya devam etmesi sağlanmıştır. Anahtar Kelimeler — GPS, GPS karıştırma, Karıştırmaya Dayanıklı GPS, Dizi Anten, Capon Hüzme Yönlendirme, Sıfır Kazanç Yönlendirme. Abstract—GPS receiver operation can be interrupted by intentional/unintentional jammers even with low power levels. As the jammers become more abundant, protection of critical systems gains importance. In this work, it is aimed to suppress the GPS L1 band jammers using spatial filtering with antenna arrays. After experimental studies, it is achieved to suppress the jammers and to recover the GPS operation in real environment.

This article puts forward a method for codebook design to support beamforming mechanism in a 60GHz millimeter-wave wireless communication environment. The codebook is designed with only phase shifting but not any amplitude adjustment to meet the low power consumption requirement. IEEE802.15.3c beam MRA (main response axis) directions are taken as the original version to make curve fitting, so that weighting vector based on uniform weighting can be obtained to generate a codebook matrix. The simulation results show that, compared with the codebook in IEEE802.15.3c, similar main lobe patterns can be obtained in the proposed codebook, while the side lobe level is lower and the antenna gain at the direction of MRA is higher. Meanwhile, the interference decreases obviously with an improved network channel capacity in the practice environment, and the codebook is also robust to the beam shifting deriving from phase errors.

A novel array signal processing technique is proposed to estimate HF channel parameters including number of paths, their respective direction of arrivals (DOA), delays, Doppler shifts and amplitudes. The proposed technique utilizes the Cross Ambiguity Function (CAF), hence, called as the CAF-DF technique. The CAF-DF technique iteratively processes the array output data and provides reliable estimates for DOA, delay, Doppler shift and amplitude corresponding to each impinging HF propagated wave onto an antenna array. Obtained results for both real and simulated data at different signal to noise ratio (SNR) values indicate the superior performance of the proposed technique over the well known MUltiple SIgnal Classification (MUSIC) technique.

A new method for estimating the direction-of-arrival (DOA) of unknown number of source signals is proposed. The angular directions of coherent and/or noncoherent signals are estimated by controlling the nulls of the antenna array factor. A genetic algorithm is used to find the nulls on the unit circle that minimize the array output power. The pseudo-spectrum is obtained by rotating the nulls on the unit circle and estimating the corresponding array output power. The simulation results indicate that the direction-of-arrivals are estimated accurately in the presence of a wide range of input noise levels.

9
th International Conference on Signal and Image Processing (Signal 2022) is a forum for
presenting new advances and research results in the fields of Digital Image Processing. The
conference will bring together leading researchers, engineers and scientists in the domain of
interest from around the world. The scope of the conference covers all theoretical and practical
aspects of the Signal, Image Processing & Pattern Recognition.

A comparison analysis between two different types of antenna arrays in direction-of-arrival (DOA) estimation is presented. This study was undertaken to point out any difference in the process of DOA estimation between isotropic and directional antenna array. To the best of our knowledge, there is no previous work that analysed the DOA estimation between these two antenna arrays. The approach taken was to estimate all predetermined angle-of-arrival (AOA) correctly by using MUSIC, a spectral-based DOA algorithm. Results obtained show that, to perform the DOA estimation, isotropic antenna array depends entirely on DOA algorithm, whereas directional antenna array required both the DOA algorithm and the position of the main beam at any instant. This work concludes that, in practical application, it is essential to consider both the performance of DOA algorithm and the beam pattern to ensure the DOA estimation is carried out correctly.

This article develops the mathematical models needed to describe the key issues in using an array of antennas for receiving spacecraft signals for DSN applications. The detrimental effects of nearby interfering sources, such as other spacecraft transmissions or natural radio sources within the array's field of view, on signal-to noise ratio (SNR) are determined, atmospheric effects relevant to the arraying problem developed, and two classes of algorithms (multiple signal classification (MUSIC) plus beam forming, and an eigen-based solution) capable of phasing up the array with maximized SNR in the presence of realistic disturbances are evaluated. It is shown that, when convolutionally encoded binary-phase shift keying (BPSK) data modulation is employed on the spacecraft signal, previously developed data pre-processing techniques that partially reconstruct the carrier can be of great benefit to array performance, particularly when strong interfering sources are present. Since this article is concerned mainly with demonstrating the required capabilities for operation under realistic conditions, no attempt has been made to reduce algorithm complexity; the design and evaluation of less complex algorithms with similar capabilities will be addressed in a future article. The performances of the candidate algorithms discussed in this article have been evaluated in terms of the number of symbols needed to achieve a given level of combining loss for different numbers of array elements, and compared on this common basis. It is shown that even the best algorithm requires approximately 25,000 symbols to achieve a combining loss of less than 0.5 dB when 128 antenna elements are employed, but generally 50,000 or more symbols are needed. This is not a serious impediment to successful arraying with high data-rate transmission, but may be of some concern with missions exploring near the edge of our solar system or beyond, where lower data rates may be required.

This paper describes recent work on the topic of multi-dimensional (MD) spatio-temporal noise and distortion shaping for radio-frequency (RF) antenna arrays with applications in wireless communications, phased-array radar sensing, microwave/mm-wave imaging, and radio astronomy instrumentation. The MD spectral properties of propagating plane-waves that arise from electromagnetics are combined with MD circuits and signal processing theory based on passive resistive terminated 2-D filters. The result, for the first time in the literature, is MD multi-port extensions of the analog and digital electronics found within wireless transceivers, including RF amplifiers, mixers, and data converters. In particular, a multi-port extension of conventional analog-to-digital converters (ADCs) is proposed in which the distortion that is generated from non-linear operations (such as quantization) is spectrally shaped in multiple spatio-temporal dimensions such that the region of support (ROS) of the desired RF signals corresponding to desired planar waves is ideally mutually exclusive with that of these undesired components. Because of spectral shaping, both distortion and noise are non-overlapping with the signal of interest and can be filtered out after sampling by using an MD digital filter. This theoretical advance in MD noise and distortion shaping across both space and time domains ensures that both electronic noise and non-linear distortion arising from coarse low-complexity quantization do not appreciably reduce the signal to noise and distortion ratio (SNDR) of antenna array receivers. The proposed method is an extension of ∆ − Σ modulation that is commonly used in conventional single-input single-output ADCs but does not require either temporal or spatial over-sampling. Moreover, to the best of our knowledge this paper is the first to combine 2-D analog filters derived from resistively-terminated classical passive low-pass filter prototypes with active analog feedback control in multiple dimensions (space, time) to realize 2-D analog-digital mixed-signal electronics for RF array processing applications.

DOA estimation methods seem to be very useful in the reconstruction of the original received signals and help on the estimation of its location which is highly applicable in radars, sonars, seismic exploration, and military surveillance. DOA estimation methods try to figure out the parameters hidden in the sensors data using different mathematical techniques and physical properties of the geometry of the array of antennas and the impinging signals themselves. In this thesis, the DOA estimation methodology of wideband signals is studied and its computational cost and execution time are investigated. Further, recent well-known wideband algorithms are also investigated and simulated. Finally, The simulation results show the main factors that affect the computational costs of DOA methods and how to control it to reduce the complexity of the algorithms while maintaining a high resolution.

The discrete time wavelet transform has been used to develop software that detects seismic P and S-phases. The detection algorithm is based on the enhanced amplitude and polarization information provided by the wavelet transform coefficients of the raw seismic data. The algorithm detects phases, determines arrival times and indicates the seismic event direction from three component seismic data that represents the ground displacement in three orthogonal directions. The essential concept is that strong features of the seismic signal are present in the wavelet coefficients across several scales of time and direction. The P-phase is detected by generating a function using polarization information while S-phase is detected by generating a function based on the transverse to radial amplitude ratio. These functions are shown to be very effective metrics in detecting P and S-phases and for determining their arrival times for low signal-to-noise arrivals. Results are compared with arrival times obtained by a human analyst as well as with a standard STA/LTA algorithm from local and regional earthquakes and found to be consistent.

Two experiments were conducted to investigate Adaptive Noise Cancelling and Cepstrum echo removal post-processing techniques on acoustic data from a linear microphone array in an anechoic chamber. A point source speaker driven with white noise was used as the primary signal. The first experiment included a background speaker to provide interference noise at three different Signal-to-Noise Ratios to simulate noise propagating down a wind tunnel circuit. The second experiment contained only the primary source and the wedges were removed from the floor to simulate reflections found in a wind tunnel environment.
The techniques were applicable to both signal microphone and array analysis. The Adaptive Noise Cancellation proved successful in its task of removing the background noise from the microphone signals at SNRs as low as -20 dB. The recovered signals were then used for array processing. A simulation reflection case was analyzed with the Cepstral technique. Accurate removal of the reflection effects was achieved in recovering both magnitude and phase of the direct signal. Experimental data resulted in Cepstral features that caused errors in phase accuracy. A simple phase correction procedure was proposed for this data, but in general it appears that the Cepstral technique is and would be not well suited for all experimental data.

This paper introduces a novel eigenstructure-based algorithm uni-vector-sensor ESPRIT that yields closed-form direction-of-arrival (DOA) estimates and polarization estimates using one electromagnetic vector sensor. A vector sensor is composed of six spatially co-located nonisotropic polarizationsensitive antennas, measuring all six electromagnetic field components of the incident wave field. Uni-vector-sensor ESPRIT is based on a matrix-pencil pair of temporally displaced data sets collected from a single electromagnetic vector sensor. The closed-form parameter estimates are obtained through a vector cross-product operation on each decoupled signalsubspace eigenvector of the data correlation matrix. This method exploits the electromagnetic sources' polarization diversity in addition to their spatial diversity, requires no a priori knowledge of signal frequencies, suffers no frequency-DOA ambiguity, pairs automatically the x-axis direction cosines with the y-axis direction cosines, eliminates array interelement calibration, can resolve up to five completely polarized uncorrelated monochromatic sources from near field or far field. It impressively out-performs an array of spatially displaced identically polarized antennas of comparable array-manifold size and computational load.

Immersive-Audio technologies are widely used to build experimental and commercial audio systems. However, most of them are based on standard PCs, which introduce performance limitations and excessive power consumption. To address these drawbacks, we explore the implementation prospectives of two Immersive-Audio technologies; the beamforming (BF) and the Wave Field Synthesis (WFS). We target two popular multicore platforms, namely Graphic Processor Units (GPUs) and Field Programmable Gate Arrays (FPGAs). We identify the most computationally intensive parts of both applications and employ the CUDA environment to map them onto a Quadro FX1700, a GeForce 8600GT, a GTX275 and a GTX460 GPU. Furthermore, we design our custom multi-core hardware accelerators for both algorithms and map them onto Virtex6 FPGAs. Both GPU and FPGA implementations are compared against OpenMP-annotated software running on a Core2 Duo at 3.0 GHz. Experimental results suggest that middle-range GPUs process data equally well as the Core2 Duo for the BF, and approximately two times faster for the WFS. However, high-end GPU and FPGA solutions provide an order of magnitude better performance for BF, and approximately two orders of magnitude better performance for WFS than the Core2 Duo. Ultimately, single-chip GPU and FPGA implementations can provide more power-effective solutions, since they can drive more complex microphone and loudspeaker setups than PC-based approaches.

This module is designed to give the students a practical understanding of performing speech enhancement using microphone arrays and demonstrate the difference between different techniques. This module is closely related to the lecture Speech Enhancement given by Prof. Dr. ir. Emanuël Habets. In this exercise, the students will implement a commonly used spatial signal processing technique known as beamforming, and analyse the performance of two different beamformers, a fixed beamformer known as delay-and-sum beamformer and a signal-dependent beamformer known as minimum variance distortionless response (MVDR) beamformer, for a noise and interference reduction task. Their performances will be compared via objective measures to demonstrate the advantages of the signal-dependent beamformers.

A strategy for adaptive control and energetic optimization of aerobic fermentors was implemented, with both air flow and agitation speed as manipulated variables. This strategy is separable in its components: control, optimization, estimation. We optimized parameter’s estimation (from the usual KLa correlation) using sinusoidal excitation of air flow and agitation speed. We have implemented parameter’s estimation trough recursive least squares algorithm with forgetting factor. We carried separate essays on control, optimization and estimation algorithms. We carried our essays using an original computational simulation environment, with noise and delay generating facilities for data sampling and filtering.

Our results show the convergence and robustness of the estimation algorithm used, improved with use of both forgetting factor and KLa dead-band facilities. Control algorithm used in our work compares favorably with PID using the integrated area criteria for deviation between oxygen molarity and critical molarity (set point). Optimization algorithm clearly reduces energetic consumption, respecting critical molarity. Integration of control, optimization and adaptive algorithms was implemented, but future work is needed for stability. Methods were defined and implemented for stability improvement. We have implemented data acquisition and computer manipulation of air flow and agitation speed for actual fermentors.

It is an experimental design problem in which there are two Poisson sources with two possible and known rates, and one counter. Through a switch, the counter can observe the sources individually or the counts can be combined so that the counter observes the sum of the two. The sensor scheduling problem is to determine an optimal proportion of the available time to be allocated toward individual and joint sensing, under a total time constraint. Two different metrics are used for optimization: mutual information between the sources and the observed counts, and probability of detection for the associated source detection problem. Our results, which are primarily computational, indicate similar but not identical results under the two cost functions.

In this study direction finding (DF) with a uniform circular array (UCA), which is placed on a moving platform, is studied. First, the problem is formulated in which the effects of both antenna array and source movements are included as Doppler phase shifts. Then, assuming that the array moves in one direction with a constant speed, a new DF algorithm is proposed. The proposed algorithm uses different time samples of the UCA’s spatial samples to construct two virtual subarrays, and therefore there is an inherent displacement between subarrays. In the proposed algorithm, the direction of arrivals (DOAs) of the incoming signals are estimated by using the sample correlation matrices of the two virtual subarrays. Thus proposed algorithm is based on the ESPRIT algorithm, and it is named as SAPESA (synthetic aperture ESPRIT algorithm). An iterative procedure is proposed for the estimation of the DOAs in both azimuth and elevation. The performance of the proposed DF algorithm is investigated through computer simulations. Its performance is compared to that of the MUSIC algorithm. It is observed that each algorithm is superior to the other in different occasions.

The evaluation of ultrasound system is measured by the development in analog and digital electronics. A modular field programmable gate array (FPGA)-based digital ultrasound beamforming is presented. The digital beamforming is implemented in Virtex-5 FPGA. The objective of this work is to develop a modular low-cost PC-based digital ultrasound imaging system that has almost all of its processing steps done on the PC side. The system consists of: two 8 channels block and reconstructed line block. The 8 channel block consist of: memory block to save the samples data after converted to fixed point type, delay block implemented by addressable shift register -the delay process is based on sampled delay focusing (SDF) -and M-code block applied the summation of each RF channel samples. The reconstructed block consists of pipelined adder to apply the summation of the two 8 channels blocks. The power consumption and device utilization was acceptable. Also it is possible to build 16-,32-,64-,and 128-channel beamformer. The hardware architecture of the design provided flexibility for beamforming.

Microphone arrays have been used in various applications to capture conversations, such as in meetings and teleconferences. In many cases, the microphone and likely source locations are known a priori, and calculating beamforming filters is therefore straightforward. In ad-hoc situations, however, when the microphones have not been systematically positioned, this information is not available and beamforming must be achieved blindly. In achieving this, a commonly neglected issue is whether it is optimal to use all of the available microphones, or only an advantageous subset of these. This paper commences by reviewing different approaches to blind beamforming, characterising them by the way they estimate the signal propagation vector and the spatial coherence of noise in the absence of prior knowledge of microphone and speaker locations. Following this, a novel clustered approach to blind beamforming is motivated and developed. Without using any prior geometrical information, microphones are first grouped into localised clusters, which are then ranked according to their relative distance from a speaker.

Vulnerability of Global Navigation Satellite System (GNSS) signals to strong radio frequency (RF) signals has resulted in new research needs. Based on the type of interference various mitigation schemes have been proposed. Antenna array processing is one of the powerful methods for interference mitigation. However, data collection using real antenna arrays to develop and test array processing methods is considerably complex. Therefore, it is very useful to develop a simpler way of obtaining array data using a simulation platform.
This paper introduces a new approach to simulate GNSS signals for an antenna array. Employing the user trajectory information recorded using a single antenna hardware simulator and the known physical configuration of the antenna array, GNSS signals are simulated for each antenna element. Additionally, directional interference signals can be generated and added to the GNSS signals. Fidelity of single antenna simulations is evaluated by comparing the tracking parameters from this simulator with two commercially available GNSS simulators, namely the Spirent’s GSS-7700 and Rohde & Schwarz’s SMBV-100. Simulations are verified by evaluating C/N0 values, carrier Doppler and code phase. Two approaches of array signal generation i.e. one based on a reference antenna and another based on independent antenna’s signal parameters, are compared. Capability of the simulator to generate interference scenarios is demonstrated with several examples and comparisons in acquisition metrics, position errors and beam patterns.

We present a novel approach for real-time multichannel speech enhancement in environments of nonstationary noise and timevarying acoustical transfer functions (ATFs). The proposed system integrates adaptive beamforming, ATF identification, soft signal detection, and multichannel postfiltering. The noise canceller branch of the beamformer and the ATF identification are adaptively updated online, based on hypothesis test results. The noise canceller is updated only during stationary noise frames, and the ATF identification is carried out only when desired source components have been detected. The hypothesis testing is based on the nonstationarity of the signals and the transient power ratio between the beamformer primary output and its reference noise signals. Following the beamforming and the hypothesis testing, estimates for the signal presence probability and for the noise power spectral density are derived. Subsequently, an optimal spectral gain function that minimizes the mean square error of the log-spectral amplitude (LSA) is applied. Experimental results demonstrate the usefulness of the proposed system in nonstationary noise environments.

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9th International Conference on Signal and Image Processing (SIGL 2022) is a forum for presenting new advances and research results in the fields of Digital Image Processing. The conference will bring together leading researchers, engineers and scientists in the domain of interest from around the world. The scope of the conference covers all theoretical and practical aspects of the Signal, Image Processing & Pattern Recognition.

For phased-array ultrasound imaging, alternative beamforming techniques and their VLSI circuits are studied to form a fully digital receive fronted hardware. This project proposes to design and develop an Ultrasound array signal processor. VLSI architecture for array signal processor would be designed and FPGA implementation will carry out. Array signal Processing is a wide area of research that extends from the simplest form of 1-D signal processing to the complex form of M-D and array signal processing. In array signal processing beam forming is the terminology used for array signal processer. Beam forming is the process of combining signals from various directions and forming a high energy beam, in a particular direction. In Ultrasound imaging various array Geometries are used depending upon various applications. Among various array geometries phased array is the most commonly used array. In this project it is proposed to design and develop a phased array beam former for 64 channels at 40 MHz sampling. Initially the algorithm will be modelled in Matlab and VLSI architecture would be developed. The VLSI architecture functionality would be compared with Matlab output. Later the performance of Architecture will be investigated for timing and area.

This paper presents a series of user parameter-free iterative Sparse Asymptotic Minimum Variance (SAMV) approaches for array processing applications based on the asymptotically minimum variance (AMV) criterion. With the assumption of abundant snapshots in the direction-of-arrival (DOA) estimation problem, the signal powers and noise variance are jointly estimated by the proposed iterative AMV approach, which is later proved to coincide with the Maximum Likelihood (ML) estimator. We then propose a series of power-based iterative SAMV approaches, which are robust against insufficient snapshots, coherent sources and arbitrary array geometries. Moreover, to overcome the direction grid limitation on the estimation accuracy, the SAMV-Stochastic ML (SAMV-SML) approaches are derived by explicitly minimizing a closed form stochastic ML cost function with respect to one scalar paramter, eliminating the need of any additional grid refinement techniques. To assist the performance evaluation, approximate solutions to the SAMV approaches are also provided for high signal-to-noise ratio (SNR) and low SNR scenarios. Finally, numerical examples are generated to compare the performances of the proposed approaches with those of the existing ones.

A mixed-signal ASIC that implements an ultrasound front-end receiver in a 0.6 m BiCMOS HotASIC technology that features metal/metal capacitors and poly1/poly2 resistors is described. The ASIC includes a low-noise amplifier (LNA), a programmable gain amplifier (PGA), an output differential amplifier (ODA), and a second-order sigma-delta modulator (SDM), and is the most compact system for high-temperature ultrasound applications reported in literature. The circuit has a programmable gain and is designed for measuring the signal response (200 kHz to 700 kHz) from an ultrasound transducer. At 48 MHz clock frequency and 200 C, the power consumption is 85 mW from a single 5 V supply. The die area of the chip is 5.52 mm 2 .

Location and identification of cooperating aircraft in the airport area (and beyond) may be implemented by multilateration (MLAT) systems using the secondary surveillance radar (SSR) mode S signals. Most of these signals, spontaneously emitted from on-board mode S transponders at a fixed carrier frequency, arrive randomly at the receiving station, as well as many mode A/C replies from legacy transponders still in use. Several SSR signals are, then, overlapped in multiple aircraft situations. Therefore, the aim of this work is the separation of overlapped SSR signals, i.e., signals superimposed in time at receiving stations. We improve the MLAT receiving station by replacing the single antenna by an array of m elements and using array signal processing techniques. In the literature, several algorithms address the general source separation problem, but a very few of them are specifically designed for a mixture of overlapping SSR replies. Unfortunately, all of them have either some shortcomings, or an expensive computational cost, or no simple practical implementation. In this paper, we use the time sparsity property of the sources to propose more reliable, simpler, and more effective algorithms based on projection techniques to separate multiple SSR signals. Real recorded signals in a live environment are used to demonstrate the effectiveness of our method.

A novel design method for lossy Blass matrix beamforming networks is presented. Compared to those formerly developed [1,2], the new method allows the design of such circuits to generate an arbitrary number of simultaneous beams oriented in arbitrary positions. This is obtained by means of an equivalent circuit approach to compute the transmission parameters of the Blass matrix. The method has been applied to the design of a specific multi-beam waveguide application, which required the development and characterization of a novel waveguide cross coupler. Experimental results confirm the accuracy of the method showing high performance for all beams in terms of beam position, side lobe level and efficiency.

In CTBT related applications Blind Source Separation (BSS) methods can be used for signal recovery from the mixture using minimal a priori information about the signals  сomposing the mixture. Expert Technical Analysis (ETA) conducted in CTBTO to improve the estimated values for the standard signal and event parameters according to the Protocol to the CTBT may face problems which cannot be resolved with certified CTBTO applications and may demand specific techniques. Here, we examine two scenarios of interest: (1) separation of two almost co-located explosions and conducted within fractions of seconds, and (2) extraction of explosion signals merged with wavetrains from strong earthquake. Independent Component Analysis (in its FastICA implementation) implying non-Gaussianity of the underlying processes signal’s mixture is a blind source separation method that we apply to resolve the mentioned above cases. We have tested this technique with synthetic waveforms, seismic data from DPRK explosions and mining blasts conducted within East-European platform as well as with signals from regional and strong teleseismic events. Our approach demonstrates a good ability of waveforms separation. We also share our experience in applying the ICA in cepstral domain for separation of seismic signals based on the finite convolution representation model.

Among Sound Source Localization (SSL) methods, beamformed microphone arrays using Steering Response Power(SRP) has received significant attention. Yet, its application is stumbled by its computational complexity which cannot meet low-power/low-cost applications requirements. In this paper, Angular beamforming is presented. By applying this beamforming approach to the conventional SRP-PHAT one can find that the complexity of SRP-PHAT will decreased by two orders of magnitudes with a mere three fold reduction in resolution. The proposed method is implemented both on FPGA and 0.18um CMOS technology for a practical case.

We investigate spatial filtering techniques for interference removal in multichannel radio astronomical observations. The techniques are based on the estimation of the spatial signature vector of the interferer from short-term spatial covariance matrices followed by a subspace projection to remove that dimension from the covariance matrix, and by further averaging. The projections will also modify the astronomical data, and hence a correction has to be applied to the long-term average to compensate for this. As shown by experimental results, the proposed technique leads to significantly improved estimates of the interference-free covariance matrix.

A new hybrid algorithm that combines the uniform circular array-RAnk REduction (UCA-RARE) and Root-MUSIC algorithm for 2-D direction-of-arrival (DOA) estimation of azimuth and elevation angle is presented for uniform circular arrays in the presence of mutual coupling. To describe mutual coupling and platform effects we rely on the circular symmetry and expand the open-circuit voltages into a limited number of phase modes. This number of phase modes only depends on the electromagnetic dimensions of the UCA and is independent of the severity of mutual coupling in the UCA. The UCA-RARE algorithm is then applied to estimate the azimuth angle independent from the elevation angle. Next, for each azimuth angle we perform a new search-free rooting algorithm based on the expansion of the array manifold into a double Fourier series. By considering several examples, it is shown that even in the presence of severe mutual coupling the proposed combined technique yields very robust DOA estimations for azimuth angle as well as for elevation angle.

The first 42 elements of the Allen Telescope Array (ATA-42) are beginning to deliver data at the Hat Creek Radio Observatory in Northern California. Scientists and engineers are actively exploiting all of the flexibility designed into this innovative instrument for simultaneously conducting panoramic surveys of the astrophysical sky. The fundamental scientific program of this new telescope is varied and exciting; we here discuss some of the first astronomical results.

Model error sensitivity is an issue common to all high-resolution direction-of-arrival estimators. Much attention has been directed to the design of algorithms for minimum variance estimation taking only finite sample errors into account. Approaches to reduce the sensitivity due to array calibration errors have also appeared in the literature. Herein, one such approach is adopted that assumes that the errors due to finite samples and model errors are of comparable size. A weighted subspace fitting method for very general array perturbation models is derived. This method provides minimum variance estimates under the assumption that the prior distribution of the perturbation model is known. Interestingly, the method reduces to the WSF (MODE) estimator if no model errors are present. Vice versa, assuming that model errors dominate, the method specializes to the corresponding "model-errors-only subspace fitting method." Unlike previous techniques for model errors, the estimator can be implemented using a two-step procedure if the nominal array is uniform and linear, and it is also consistent even if the signals are fully correlated. The paper also contains a large sample analysis of one of the alternative methods, namely, MAPprox. It is shown that MAPprox also provides minimum variance estimates under reasonable assumptions.

A millimeter-wave (mm-wave) impulse radio is an attractive alternative to existing high-speed mm-wave radio systems because of the potential for a simpler transceiver architecture. This paper studies the feasibility of a mm-wave impulse radio system for home and office use by considering state-ofthe-art transceivers and multiple-input multiple-output measured propagation channels as well as the IEEE 802.15.3c channel model. Our analysis reveals that reliable data transmission is infeasible even in line-of-sight (LOS) scenarios because of a low power level at the Rx if we do not use beamforming. However, introducing 7 × 7 beamforming at the Tx dramatically improves the coverage beyond 5 m distance in the LOS scenario and up to 5 m in a non-line-of-sight (NLOS) scenario, though the performance varies in the NLOS scenario, depending on the Tx and Rx locations. We propose an adaptive transmit signaling scheme that adjusts the pulse repetition frequency depending on the delay dispersion of the propagation channel in order to avoid intersymbol interference and keep the Rx structure simple. The proposed transmit signaling scheme leads to a pulse transmission rate of 250 Mpulses/s in all measured channels while the rate is lower when the 802.15.3c model is considered because of a more multipath-rich characteristics than our measured channels.