This investigation proposed two array beamformers SPFDBB (Soft Penalty Frequency Domain Block Beamformer) and FDABB (Frequency Domain Adjustable Block Beamformer). Compared with the conventional beamformers, these frequency-domain methods can significantly reduce the computation power requirement in ASR (Automatic Speech Recognition) based applications. Like other reference signal based techniques, SPFDBB and FDABB minimize microphone's mismatch, desired signal cancellation caused by reflection effects and resolution due to the array's position. Additionally, these proposed methods are suitable for both near-field and far-field environments. Generally, the convolution relation between channel and speech source in time domain cannot be modeled accurately as a multiplication in the frequency domain with a finite window size, especially in ASR applications. SPFDBB and FDABB can approximate this multiplication by treating several frames as a block to achieve a better beamforming result. Moreover, FDABB adjusts the number of frames on-line to cope with the variation of characteristics in both speech and interference signals. A better performance was found to be achievable by combining these methods with an ASR mechanism.

This paper proposes the design to reduce the number of fed elements by replacing with parasitic elements in an array antenna. The study depends on the analysis of electromagnetic wave fields in consideration of the coupling between the half-wavelength dipoles. The case of 2 fed elements and 2 parasitic elements is considered as a unit cell to form the total array. After optimizing the element arrangement, the antenna gain can match that of the equivalent 4-fed element case. Feeding networks in a high power radiating system are analyzed in terms of the length and matching of feed lines, and the arrangement of amplifiers.

It is interesting to resolve coherent signals impinging upon a linear sensor array with low computational complexity in array signal processing. In this paper, a computationally efficient method of signal subspace fitting (SSF) for direction-of-arrival (DOA) estimation is developed, based on the multi-stage wiener filter (MSWF). To find the new signal subspace, the proposed method only needs to compute the matched filters in the forward recursion of the MSWF, does not involve the estimate of an array covariance matrix or any eigendecomposition, thus implying that the proposed method is computationally efficient. Numerical results show that the proposed method provides the comparable estimation accuracy with the classical weighted subspace fitting (WSF) method for uncorrelated signals at reasonably high SNR and reasonably large samples, and surpasses the latter for coherent signals in the case of low SNR and small samples. When SNR is low and the samples are small, the proposed method is less accurate than the classical WSF method for uncorrelated signals. This drawback is balanced by the computational advantage of the proposed method.

A compensation method of the array element pattern is proposed to measure EM field distribution on an observation plane located several wavelengths away from electronic devices in a short time. Numerical and experimental data of the 3 and 5 element collinear dipole array sensors are presented to demonstrate the validity of the proposed method.

This paper proposes near-field sound-source localization based on crosscorrelation of a signed binary code. The signed binary code eliminates multibit signal processing for simpler implementation. Explicit formulae with near-field assumption are derived for a two microphone scenario and extended to a three microphone case with front-rear discrimination. Adaptive threshold for enabling and disabling source localization is developed for robustness in noisy environment. The proposed sound-source localization algorithm is implemented on a fixed-point DSP. Evaluation results in a robot scenario demonstrate that near-field assumption and front-rear discrimination provides almost 40% improvement in DOA estimation. A correct detection rate of 85% is obtained by a robot in a home environment.

This paper considers the problem of finding two-dimensional (2-D) direction of arrivals (DOAs) for coherent cyclostationary signals using a 2-D array with random position errors. To alleviate the performance degradation due to the coherence between the signals of interest (SOIs) and the random perturbation in 2-D array positions, a matrix reconstruction scheme in conjunction with an iterative algorithm is presented to reconstruct the correlation matrices related to the received array data so that the resulting correlation matrices possess the eigenstructures required for finding 2-D DOAs. Then, using the reconstructed matrices, we create a subspace orthogonal to the subspace spanned by the direction vectors of the SOIs. Therefore, the 2-D DOAs of the SOIs can be estimated based on a subspace-fitting concept and the created subspace. Finally, several simulation examples are presented for illustration and comparison.

In this paper, we propose a novel array antenna-assisted adaptive modulation scheme for fast fading environments. Although adaptive modulation is an efficient technique capable of establishing high bit-rate digital transmission in a multi-path fading environment, it is sensitive to the fast time variation of the channel because of difficulties in tracking the channel state. To resolve this problem, an array antenna-based Doppler spread compensator was applied to the adaptive modulation scheme. Computer simulation results indicated that the proposed scheme can markedly improve the bit error rate and throughput performance for the region in which the maximum Doppler frequency normalized by the packet length is up to 0.1.

We present a low-phase-noise and frequency-tunable photonic millimeter-wave (MMW) generator based on two-mode beating. The generator consists of a single-mode laser, an external optical intensity modulator, and a planar lightwave circuit (PLC) on which an arrayed-waveguide grating (AWG) and 3-dB optical combiners are integrated. Because the AWG and the optical combiners are connected with optical waveguides and the optical path length difference between the two modes filtered by the AWG is kept constant, the phase fluctuation of the generated MMW signal is suppressed. The generator can generate MMWs with a phase noise of less than -75 dBc/Hz at 100 Hz and has a frequency tunability in a range of 90 to 125 GHz. The generator can be applied for the local oscillator (LO) in 10-Gbit/s wireless links that use heterodyne detection.

This paper introduces the multiple signal classification (MUSIC) method that utilizes the transfer characteristics of microphones located at the same place, namely aggregated microphones. The conventional microphone array realizes a sound localization system according to the differences in the arrival time, phase shift, and the level of the sound wave among each microphone. Therefore, it is difficult to miniaturize the microphone array. The objective of our research is to build a reliable miniaturized sound localization system using aggregated microphones. In this paper, we describe a sound system with N microphones. We then show that the microphone array system and the proposed aggregated microphone system can be described in the same framework. We apply the multiple signal classification to the method that utilizes the transfer characteristics of the microphones placed at a same location and compare the proposed method with the microphone array. In the proposed method, all microphones are placed at the same place. Hence, it is easy to miniaturize the system. This feature is considered to be useful for practical applications. The experimental results obtained in an ordinary room are shown to verify the validity of the measurement.

A robust microphone array for speech enhancement and noise suppression is studied in this paper. To overcome target signal cancellation problem of conventional beamformer caused by array imperfections or reverberation effects of acoustic enclosure, the proposed microphone array adopts an arbitrary model of channel transfer function (TF) relating microphone and speech source. Since the estimation of channel TF itself is often intractable, herein, transfer function ratio (TFR) is estimated instead and used to form a suboptimal beamformer. A robust TFR estimation method is proposed based on signal subspace analysis technique against stationary or slowly varying noise. Experiments using simulated signal and actual signal recorded in a real room illustrate that the proposed method has high performance in adverse environment.

This paper overviews a total solution for frequency-domain blind source separation (BSS) of convolutive mixtures of audio signals, especially speech. Frequency-domain BSS performs independent component analysis (ICA) in each frequency bin, and this is more efficient than time-domain BSS. We describe a sophisticated total solution for frequency-domain BSS, including permutation, scaling, circularity, and complex activation function solutions. Experimental results of 22, 33, 44, 68, and 22 (moving sources), (#sources#microphones) in a room are promising.

In this letter, we propose a closed-form joint space-time channel and direction of arrival (DOA) blind estimation algorithm for space-time coded MC-CDMA systems equipped with a uniform linear array (ULA) at the base station in frequency-selective fading environments. We prove that the signal subspaces defined by the receive data covariance matrix can be determinately separated into an equivalent set of signal subspaces spanned by the space-time channel vector of an individual user. From these signal subspaces, the space-time channels of multiple users are estimated using the subspace method.

Mainly, a uniform linear array (ULA) has been used for DOA estimation of coherent signals because we can apply the spatial smoothing preprocessing (SSP) technique. However, estimation by a ULA has ambiguity due to the symmetry, and the estimation accuracy depends on the DOA. Although these problems can be solved by using a uniform circular array (UCA), we cannot estimate the DOA of coherent signals because the SSP technique cannot be applied directly to the UCA. In this paper, we propose to estimate 2-dimensional DOA (polar angles and azimuth angles) estimation of coherent signals using a cylindrical array which is composed of stacked UCAs.

A systolic array is an ideal for ASICs because of its massive parallelism with a minimum communication overhead, regularity and modularity. Most of commercial FPGAs cannot handle systolic structure with fast sampling rate for their general-purpose architecture nature. This paper presents a new PLD architecture targeting a super-systolic array for application-specific arithmetic operations such as MAC. This architecture combines the high performance of ASICs with the flexibility of PLDs and it offers a significant alternative view on the programmable logic devices. The super-systolic array is ideal for a newly proposed PLD architecture when it comes to area-efficiency, P&R and clock speed.

We propose an information-sharing network system, capable of forming and dynamically reconfiguring multiple information-sharing groups on the same network platform by using wavelength routing and distributed shared memory technologies. The network system comprises information-sharing terminal nodes equipped with a shared memory and a wavelength-tunable transmitter, network management terminal and an arrayed-waveguide grating (AWG). The information-sharing terminal nodes are connected to an AWG by a pair of optical fibers, forming a star-shaped topology. Information is shared among the information-sharing terminal nodes through the establishment of a logical information-sharing ring. This is accomplished by adjusting the output of the wavelength-tunable transmitter at each terminal node to an appropriate wavelength according to the wavelength-routing characteristics of the AWG wavelength router. We developed a prototype information-sharing network system, in which, as preliminary experiments, HDTV and SDTV videos were used for real-time information sharing. The dynamic reconfiguration of information-sharing groups and a simple automatic restoration technique have been successfully demonstrated. The system is applicable to distributed computer processing systems and high-capacity information-sharing applications such as high-quality videoconferences.

The reflection characteristics of large alternating-phase fed single-layer waveguide arrays with center-feeds are investigated to identify the mechanism for bandwidth narrowing effects. Firstly, the overall reflection for the whole array is analyzed by FEM and fine agreement with measurements is demonstrated. It is deviating from the conventional prediction based upon a simple sum of reflections from components in the array, such as the multiple-way power divider, the slot waveguides and the aperture at the antenna input. Careful diagnosis reveals that the mutual coupling between the alternating phase waveguides via external half-space is the key factor in reflection accumulation. Amongst all, the slot with strong excitation whose position depends upon the aperture illumination design produces the dominant contribution in the mutual coupling.

This paper describes a new interface for a barge-in free spoken dialogue system combining an adaptive sound field control and a microphone array. In order to actualize robustness against the change of transfer functions due to the various interferences, the barge-in free spoken dialogue system which uses sound field control and a microphone array has been proposed by one of the authors. However, this method cannot follow the change of transfer functions because the method consists of fixed filters. To solve the problem, we introduce a new adaptive sound field control that follows the change of transfer functions.

Traveling electromagnetic waves on infinite linear periodic arrays of lossless penetrable spheres can be conveniently analyzed using the source scattering-matrix framework and vector spherical wave functions. It is assumed that either the spheres are sufficiently small, or the frequency such, that the sphere scattering can be treated using only electric and magnetic dipole vector spherical waves, the electric and magnetic dipoles being orthogonal to each other and to the array axis. The analysis simplifies because there is no cross-coupling of the modes in the scattering matrix equations. However, the electric and magnetic dipoles in the array are coupled through the fields scattered by the spheres. The assumption that a dipolar traveling wave along the array axis can be supported by the array of spheres yields a pair of equations for determining the traveling wave propagation constant as a function of the sphere size, inter-sphere separation distance, the sphere permittivity and permeability, and the free-space wave number. These equations are obtained by equating the electric (magnetic) field incident on any sphere of the array with the sum of the electric (magnetic) fields scattered from all the other spheres in the array. Both equations include a parameter equal to the ratio of the unknown normalized coefficients of the electric and magnetic dipole fields. By eliminating this parameter between the two equations, a single transcendental equation is obtained that can be easily solved numerically for the traveling wave propagation constant. Plots of the k - β diagram for different types and sizes of spheres are shown. Interestingly, for certain spheres and separations it is possible to have multiple traveling waves supported by the array. Backward traveling waves are also shown to exist in narrow frequency bands for arrays of spheres with suitable permittivity and permeability.

We have proposed a novel spatial filtering technique named "VIrtual Subcarrier Assignment (VISA)" for orthogonal frequency division multiplexing (OFDM) signals, which enables the transceiver equipped with an adaptive array antenna (AAA) to selectively receive or reject OFDM signals through coloring them with different virtual subcarrier positions in their frequency spectra. In this paper, we develop the VISA to use multiple virtual subcarrier assignment, which assigns a different combination of multiple virtual subcarrier positions in the frequency spectrum to each OFDM signal. Furthermore, we present two kinds of recursive least square (RLS)-based array weight control methods to support the VISA with multiple subcarrier puncturing in an IEEE802.11a-based system and evaluate the link-level performance in typical indoor wireless environments by computer simulations.

Convergence of the iterative method based on the successive overrelaxation (SOR) method is investigated to solve the matrix equation in the moment analysis of array antennas. It is found this method can be applied to the sub domain method of moments with fast convergence if the grouping technique is applied and the over-relaxation parameter is properly selected, and the computation time for solving the matrix equation can be reduced to be almost proportional to the second power of the number of unknowns.