This letter develops a practical sectorized antenna array using center-fed half-wavelength dipole antennas that are parallel to and a distance in front of a large ground plane reflector. Each element in the array is designed to provide coverage to isolate each 120sector from adjacent sectors. We derive a closed-form expression for spatial correlation function that can be used as guides in evaluating the effects of array spatial correlation on diversity performance in sectorized cellular communications.

A double surface light emission backlight that uses single light-guide plate, has been developed for illumination of two liquid-crystal displays (LCD) on its front and rear, to be used in a cellular phone. The light-guide plate has a trapezoid cross-section with arrays of optical micro deflector and micro prism on the front and the rear surfaces, respectively. Propagated light, forward and backward, inside the light-guide plate are controlled and directed toward LCDs using only two prism sheets with internal reflection characteristic, each for the front and the rear. Only three optical components and four light-emitting diodes (LEDs) are used in the new structure compared with ten components and six LEDs of the current type. Comparing with the current type, the thickness and power consumption of the new backlight are reduced by a factor of 0.59 and 0.67, respectively.

This paper proposes a simple method for estimation and compensation of signal direction, to deal with relative change of sound source location caused by the movements of a microphone array and a sound source. This method introduces a delay filter that has shifted and sampled sinc functions. This paper presents a concept for the joint optimization of arrival time differences and of the coordinate system of a mobile microphone array. We use the LMS algorithm to derive this method by maintaining a certain relationship between the directions of the microphone array and the sound source directions. This method directly estimates the relative directions of the microphone array to the sound source directions by minimizing the relative differences of arrival time among the observed signals, not by estimating the time difference of arrival (TDOA) between two observed signals. This method also compensates the time delay of the observed signals simultaneously, and it has a feature to maintain that the output signals are in phase. Simulation results support effectiveness of the method.

Some conventional beamformers require the direction of the desired signal. The performance of such beamformers can substantially be degraded even in the presence of small error on the directional information. In this letter, we propose a prefilter-type beamforming scheme robust to directional error by employing a simple compensator. The performance of the proposed scheme is verified by computer simulation.

Efficient schemes to enhance the performance of the optimum beamforming for DS/CDMA systems are proposed. The main focus of the proposed schemes is to enhance the practical estimation of an array response vector used at the weight vector for the optimum beamforming. The proposed schemes for the performance enhancement of the optimum beamforming are the Complex Toeplitz Approximation (CTA) and the real Toeplitz-plus-Hankel Approximation (RTHA) which have the theoretical property of an overall noise-free signal. It is shown through several simulation results that the performance of the optimum beamforming using the proposed schemes is much superior to that of a system using the conventional method under several simulation environments, i.e., the number of users, the SNR value, the number of antenna elements, the angular spread, and Nakagami fading parameter.

We present a built-in self-reconfiguring system for a mesh-connected processor array where faulty processor elements are compensated for by spare processing elements located in one row and one column. It has advantages in that the number of spare processing elements is small and additional control circuits and networks for changing interconnections of processing elements is so simple that hardware overhead for reconfiguration is also small. First, to indicate the motivation to the proposed reconfiguration scheme, we briefly describe other schemes with the same number of spares as that of the proposed scheme where faulty processing elements are replaced using straight shifts toward spares, and compare their reconfiguration probabilities to each other. Then, we show that a variant of the proposed scheme has the highest probability. Next, we present a built-in self-reconfiguring system for the scheme and formally prove that it works correctly. It can automatically replace faulty processors by spare processors on detecting faults of processors.

A novel signal enhancement scheme using the rotation of signal subspace (RSS) and Toeplitz matrix approximation (TMA) methods to enhance the performance of an adaptive antenna array in multirate DS/CDMA systems is proposed. The basis of RSS is to find a transformation matrix in order to recover the desired complex array covariance matrix from a sampled complex array covariance matrix which is contaminated by an interference-plus-noise component, which is the total noise. Also, the objective of TMA is to change the output matrix of RSS into a matrix having the theoretical properties of a total noise-free signal. Consequently, the proposed signal enhancement scheme using RSS and TMA methods can greatly improve the performance of an adaptive antenna array by reducing the undesired total noise effect from the sampled complex array covariance matrix of the pre-correlation received signal vector that is used to calculate a weight vector of an adaptive antenna array. It is shown through various simulation results that the system performance using the proposed signal enhancement scheme is much superior to that of the conventional method.

In this letter, we investigate the performance of using subband adaptive loading for the combination of orthogonal frequency division multiplexing (OFDM) and adaptive antenna array. The frequency-domain adaptive loading is very effective to deal with the frequency-selective fading which is inevitable in broadband wireless communications. However, almost all of the existing adaptive loading algorithms are based on "subcarrier-to-subcarrier" mode which may results in awfully large signaling overhead, since every subcarrier needs its own signaling loop between the transmitter and receiver. We investigate the performance of the combination of OFDM and adaptive antenna array when a subband adaptive loading algorithm is used to decrease the signaling overhead. It is shown by simulation results that at the cost of some tolerable performance loss, the signaling overhead of adaptive loading can be greatly reduced.

In this paper, we propose a method for constructing quasi-cyclic low-density parity-check codes randomly using cyclic shift submatrices on the basis of the girth of the Tanner graphs of these codes. We consider (3, K)-regular codes and first derive the necessary and sufficient conditions for weight-4 and weight-6 codewords to exist. On the basis of these conditions, it is possible to estimate the probability that a random method will generate a (3, K)-regular code with a minimum distance less than or equal to 6, and the proposed method is shown to offer a lower probability than does conventional random construction. Simulation results also show that it is capable of generating good codes both regular and irregular.

We describe elements of a fast integral equation solver for large periodic and partly periodic finite array systems. A key element of the algorithm is utilization (in a rigorous way) of a block-Toeplitz structure of the impedance matrix in conjunction with either conventional Method of Moments (MoM), Fast Multipole Method (FMM), or Fast Fourier Transform (FFT)-based Adaptive Integral Method (AIM) compression techniques. We refer to the resulting algorithms as the (block-)Toeplitz-MoM, (block-)Toeplitz-AIM, or (block-)Toeplitz-FMM algorithms. While the computational complexity of the Toeplitz-AIM and Toeplitz-FMM algorithms is comparable to that of their non-Toeplitz counterparts, they offer a very significant (about two orders of magnitude for problems of the order of five million unknowns) storage reduction. In particular, our comparisons demonstrate, that the Toeplitz-AIM algorithm offers significant advantages in problems of practical interest involving arrays with complex antenna elements. This result follows from the more favorable scaling of the Toeplitz-AIM algorithm for arrays characterized by large number of unknowns in a single array element and applicability of the AIM algorithm to problems requiring strongly sub-wavelength resolution.

A readout circuit involving new two step current mode background suppression is studied for 2-dimensional long wavelength infrared focal plane arrays (LWIR FPA's). Buffered direct injection (BDI) and feedback amplifier structure are adopted for input circuit and background suppression circuit, respectively. The pixel circuit is simple and has very small skimming error less than 0.1%. Enough calibration range over 50% as well as long integration time over 1.75 ms can be obtained using this readout circuit.

For future high-speed wireless communications using orthogonal frequency division multiplexing (OFDM), two major system requirements will emerge: throughput improvement and rich interference elimination. Because of its broadband nature and limited frequency allocations worldwide, interference from co-located wireless LAN's operating in the same frequency band will become a serious deployment issue. Adaptive array antenna can enhance the performance by suppressing the co-channel interference even when interference may have a large amount of multipath and also have similar received power to the desired signal. There are typically two types of adaptive array architecture for OFDM systems, whose signal processing is carried out before or after FFT (Fast Fourier Transform). In general, the pre-FFT array processing has low complexity, but in rich multipath and interference environments, the performance will deteriorate drastically. In contrast, the post-FFT array processing can provide the optimum performance even in such severe environments at the cost of complexity. Therefore, complexity-reduction techniques combined with the achievement of high system performance will be a key issue for adaptive array antenna applications. This paper proposes novel adaptive array architecture, which is a complexity-reduction technique using subcarrier clustering for post-FFT adaptive array. In the proposed scheme, plural subcarriers can be clustered into a group with the same spatial weight. Simulation results show that the proposed architecture is a promising candidate for real implementation, since it can achieve high performance with much lower complexity even in a rich multipath environment with low signal to noise plus interference ratio (SNIR).

A directional audible sound can be generated by amplitude-modulated (AM) into ultrasound wave from a parametric array. To synthesize audio signals produced by the self-demodulation effect of the AM sound wave, a quasi-linear analytical solution, which describes the nonlinear wave propagation, is developed for fast numerical evaluation. The radiated sound field is expressed as the superposition of Gaussian Beams. Numerical results are presented for a rectangular parametric loudspeaker, which are in good agreement with the experimental data published previously.

First we introduce the high-frequency equivalent circuit model of the Fine Pitched Ball Grid Array (FPBGA) bonding for frequencies up to 20 GHz. The lumped circuit model of the FPBGA bonding was extracted based on S-parameters measurement and subsequent fitting of the model parameters. The test packages, which contain probing pads, coplanar waveguides and FPBGA ball bonding, were fabricated and measured. The suggested π-model of the FPBGA bonding consists of self-inductor, self-capacitor, and self-resistor components. From the extracted model, a solder ball of 350 µm diameter and 800 µm ball pitch has less than 0.08 nH self-inductance, 0.40 pF self capacitance, and about 10 mΩ self-resistance. In addition, the mutual capacitance caused by the presence of the adjacent bonding balls is included in the model. The FPBGA solder ball bonding has less than 1.5 dB insertion loss up to 20 GHz, and it causes negligible delay time in digital signal transmission. The extracted circuit model of FPBGA bonding is useful in design and performance simulation of advanced packages, which use FPBGA bonding.

Integrated implementation of RF front-end components has been shown to posses many benefits. Furthermore, it presents a new way of approaching RF design. This paper will discuss the recent developments by the author's group in the field of RF front-end technology. This will include stand-alone RF front-end components such as a self-heterodyne mixer as well as more functional front-end circuitry such as digital beamformer arrays, retrodirective arrays and an array error calibration scheme.

We newly propose a novel blind separation framework for Single-Input Multiple-Output (SIMO)-model-based acoustic signals using an extended ICA algorithm, SIMO-ICA. The SIMO-ICA consists of multiple ICAs and a fidelity controller, and each ICA runs in parallel under the fidelity control of the entire separation system. The SIMO-ICA can separate the mixed signals, not into monaural source signals but into SIMO-model-based signals from independent sources as they are at the microphones. Thus, the separated signals of SIMO-ICA can maintain the spatial qualities of each sound source. In order to evaluate its effectiveness, separation experiments are carried out under both nonreverberant and reverberant conditions. The experimental results reveal that the signal separation performance of the proposed SIMO-ICA is the same as that of the conventional ICA-based method, and that the spatial quality of the separated sound in SIMO-ICA is remarkably superior to that of the conventional method, particularly for the fidelity of the sound reproduction.

The DAPDNA®-2 is the world's first general purpose dynamically reconfigurable processor for commercial usage. It is a dual-core processor consisting of a custom RISC core called the Digital Application Processor (DAP), and a two dimensional array of dynamically reconfigurable processing elements referred to as the Distributed Network Architecture (DNA). The DAP has a 32 bit instruction set architecture with an 8 KB instruction cache and 8 KB data cache that can be accessed in one clock cycle. It has an interrupt control function to detect data processing completion in the DNA-Matrix. The DNA-Matrix has different types of data processing elements such as ALU, delay, and memory elements to process fully parallel computations. The DNA-Matrix includes 32 independent 16 KB high speed SRAM elements (in total 512 KB). The DNA-Matrix, even with its parallel computational capability, can be synchronized and co-work at the same clock frequency as the DAP. The processor operates at a 166 MHz working frequency and fabricated with a 0.11 µm CMOS process. The DAPDNA-2 device can be connected directly with up to 16 units with linear scalability in processing performance, provided the bandwidth requirement is within the maximum communication speed between DNAs, which is 32 Gbps. The DAPDNA-2 performs at a level that is two orders of magnitude higher than conventional high performance processors.

In this paper the correlation spectrum of antenna array is introduced. Based on the relationship between the correlation spectrum and space spectrum of MUSIC, we proposed a novel approach to improve the DOA estimation by arranging the linear antenna array elements using genetic algorithm (GA) in optimizing the correlation spectrum. The DOA estimation performance of the optimized array is validated by Monte Carlo simulation and Cramer-Rao bound (CRB), which are improved compared with that of the traditional uniform linear array and the Minimum-Redundancy array (MRA).

In this paper, the design of a mobile satellite Internet access (MSIA) system and a mobile broadband satellite access system, called Mobile Broadband Interactive Satellite Multimedia Access Technology System (MoBISAT) are presented. MSIA system provides Internet service, broadcasting, and digital A/V service in both fixed and mobile environments using Ku-band geostationary earth orbit (GEO) satellite. A Ku-band two-way active phased array antenna installed on top of the transportation vehicles can enable the transmission of signals to satellite as well as signal tracking and reception. The forward link and return link are a high speed Time Division Multiplex (TDM) and TDMA transmission media, respectively, both of which carry signaling and user traffic. The MoBISAT, which is a next generation mobile broadband satellite access system, provides both Ku-band satellite TV and Ka-band high-speed Internet to the passengers and crews for land, maritime, and air vehicles. This paper addresses the main technological solutions adopted for the implementation and test results for the MSIA system and the main design features of the MoBISAT system.

A simple millimeter-wave quasi-maximal-ratio-combin-ing antenna diversity system based on the millimeter-wave self-heterodyne transmission technique is described. The millimeter-wave self-heterodyne transmission technique is useful for developing millimeter-wave systems with enhanced characteristics in regard to system miniaturization, development and fabrication cost, and the frequency stability of the signal transmission. We also show that applying this technique with an antenna diversity receiver configuration can easily solve a problem peculiar to millimeter-wave systems--the fact that the transmission link always requires a line-of-sight path--without requiring hardware designed with millimeter-scale precision. In this paper, we theoretically analyze the operating principle of a combining antenna diversity system based on the millimeter-wave self-heterodyne transmission technique. We further prove that we can obtain a diversity gain in accordance with that of a maximal-ratio combining diversity system without resorting to any complicated control of the received signal envelope and phase. Our experiments using the simplest two-branch diversity structure have validated the operating principle derived in our theoretical analysis. Our results show that a received CNR improvement of 3 dB is obtained as a diversity gain. We also demonstrate that circuit precision corresponding to the wavelength of the intermediate frequency, rather than to the millimeter wavelength, is sufficient to obtain the diversity effect when we control the signal phase or delay in combining the received signals.