A dual-modulus (divide-by-128/129) prescaler has been designed based on 0.25-µm CMOS technology employing new D-flip-flops. The new D-flip-flops are free from glitch problems due to internal charge sharing. Transistor merging technique has been employed to reduce the number of transistors and to secure reliable high-speed operation. At the 2.5-V supply voltage, the prescaler using the proposed dynamic D-flip-flops can operate up to the frequency of 2.95-GHz, and consumes about 10% and about 27% less power than Yuan/Svensson's and Huang's circuits, respectively.

Invention and development of the Yagi-Uda antenna, and the self-complementary antenna are described. Analysis methods of large loop antennas and the improved circuit theory (ICT) for design of linear antennas are presented. Recent developments of axial mode helical antennas and spiral antennas for radiating circularly polarized waves are also described.

This paper reviews antenna technologies for cellular phone terminal in Japan. In regard to the technologies two technical subjects are addressed: a diversity system and miniaturization of terminal size. These subjects provide motivation for studies on design methods for diversity antennas, development of built-in antennas, and control methods for surface current on terminal cavity. These studies are progressing.

This paper reviews the historical development of adaptive antennas in Japan. First of all, we watch basic adaptive algorithms. In 1980s, particularly, the following issues were a matter of considerable concern to us; (a) behavior to the coherent interference like multipath waves or radar clutters, (b) signal degradation in case that the direction of arrival (DOA) of desired signal is different from the DOA specified beforehand in the adaptive antennas with the DOA of the desired signal as a prior knowledge, and (c) performance of adaptive antennas when the desired signal and interference are broadband. Although there are a lot of development and modification of adaptive algorithms in Japan, we refer in this paper only to the above-mentioned topics. Secondly, our attention is paid to implementation of adaptive antennas and advanced technologies. A large number of researches on the subjects have been carried out in Japan. Particularly, we focus on the initiative studies in Japan toward mobile communication application. They include researches of mobile radio propagation for adaptive antennas, calibration methods, and adaptive antenna for mobile terminals. As a matter of course, we also refer to adaptive antenna technologies for advanced communication schemes such as CDMA, SDMA, OFDM and so on. Finally, we take notice of some pilot products which were developed to verify the effect of the adaptive antenna in the practical environments. As the initiative ones, a couple of equipments are introduced in this paper.

It is well known that KT method proposed by R. Kumaresan and D. W. Tufts is used as a popular parameter estimation method of exponentially damped signal. It is based on linear backward-prediction method and singular value decomposition (SVD). However, it is difficult to estimate parameters correctly by KT method in the case when high noise exists in the signal. In this paper, we propose a parameter (frequency components and damping factors) estimation method to improve the performance of KT method under high noise. In our proposed method, we find the signal zero groups by calculating zeros with different data record lengths according to the combination of forward-prediction and backward-prediction, the mean value of the zeros in the signal zero groups are calculated to estimate the parameters of the signal. The proposed method can estimate parameters correctly and accurately even when high noise exists in the signal. Simulation results are shown to confirm the effectiveness of the proposed method.

In this paper, a novel confidence scoring method that is applied to N-best hypotheses (word candidates) output from an HMM-based classifier is proposed. In the first pass of the proposed method, the HMM-based classifier with monophone models outputs N-best hypotheses and boundaries of all monophones in the hypotheses. In the second pass, an SM (Subspace Method)-based verifier tests the hypotheses by comparing confidence scores. To test the hypotheses, at first, the SM-based verifier calculates the similarity between phone vectors and an eigen vector set of monophones, then this similarity score is converted into a likelihood score with normalization of acoustic quality, and finally, an HMM-based likelihood of word level and an SM-based likelihood of monophone level are combined to formulate the confidence measure. Two kinds of experiments were performed to evaluate this confidence measure on speaker-independent word recognition. The results showed that the proposed confidence scoring method significantly reduced the word error rate from 4.7% obtained by the standard HMM classifier to 2.0%, and in an unknown word rejection, it reduced the equal error rate from 9.0% to 6.5%.

In this paper we consider blind source separation (BSS) problem of signals which are spatially uncorrelated of order four, but temporally correlated of order four (for instance speech or biomedical signals). For such type of signals we propose a new sufficient condition for separation using fourth order statistics, stating that the separation is possible, if the source signals have distinct normalized cumulant functions (depending on time delay). Using this condition we show that the BSS problem can be converted to a symmetric eigenvalue problem of a generalized cumulant matrix Z(4)(b) depending on L-dimensional parameter b, if this matrix has distinct eigenvalues. We prove that the set of parameters b which produce Z(4)(b) with distinct eigenvalues form an open subset of RL, whose complement has a measure zero. We propose a new separating algorithm which uses Jacobi's method for joint diagonalization of cumulant matrices depending on time delay. We empasize the following two features of this algorithm: 1) The optimal number of matrices for joint diago- nalization is 100-150 (established experimentally), which for large dimensional problems is much smaller than those of JADE; 2) It works well even if the signals from the above class are, additionally, white (of order two) with zero kurtosis (as shown by an example).

An adaptive array antenna for the suppression of high-power interference in direct-sequence code-division multiple access (DS-CDMA) systems is presented. Although DS-CDMA has sufficient flexibility to support a variety of services, from voice to moving-pictures, with high levels of quality, multiple access interference (MAI) is a problem. This is particularly so of the high-power interference which accompanies high-speed transmission in DS-CDMA. While the application of adaptive array antennas is an effective way of improving signal-to-interference-plus-noise ratio (SINR), problems with this approach include large levels of power consumption and the high costs of hardware and of implementing the antennas. Therefore, our main purpose is to realize a simple configuration for an adaptive array system. In order to reduce the required amounts of processing, a common beam provides suppression of high-power interference for the low-bit-rate users; this makes per-user preparation of weights unnecessary. This approach also reduces the consumption of power by the system. Interference is cancelled by minimization of the array output power (i.e., the application of a power inversion algorithm) before despreading. The approach also allows us to improve the implementation of the antenna elements by using small auxiliary antennas. The basic performance of the system is confirmed through numerical calculation and computer simulation. Furthermore, a real-time processing unit has been developed and the effectiveness of the approach is confirmed by an experiment in a radio-anechoic chamber.

The present paper deals with the blind deconvolution of a Multiple-Input Multiple-Output Finite Impulse Response (MIMO-FIR) system. To deal with the blind deconvolution problem using the second-order statistics (SOS) of the outputs, Hua and Tugnait considered it under the conditions that a) the FIR system is irreducible and b) the input signals are spatially uncorrelated and have distinct power spectra. In the present paper, the problem is considered under a weaker condition than the condition a). Namely, we assume that c) the FIR system is equalizable by means of the SOS of the outputs. Under b) and c), we show that the system can be blindly identified up to a permutation, a scaling, and a delay using the SOS of the outputs. Moreover, based on this identifiability, we show a novel necessary and sufficiently condition for solving the blind deconvolution problem, and then, based on the condition, we propose a new algorithm for finding an equalizer using the SOS of the outputs, while Hua and Tugnait have not proposed any algorithm for solving the blind deconvolution under the conditions a) and b).

Blind separation of sources from their linear mixtures is a well understood problem. However, if the mixtures are nonlinear, this problem becomes generally very difficult. This is because both the nonlinear mapping and the underlying sources must be learned from the data in a blind manner, and the problem is highly ill-posed without a suitable regularization. In our approach, multilayer perceptrons are used as nonlinear generative models for the data, and variational Bayesian (ensemble) learning is applied for finding the sources. The variational Bayesian technique automatically provides a reasonable regularization of the nonlinear blind separation problem. In this paper, we first consider a static nonlinear mixing model, with a successful application to real-world speech data compression. Then we discuss extraction of sources from nonlinear dynamic processes, and detection of abrupt changes in the process dynamics. In a difficult test problem with chaotic data, our approach clearly outperforms currently available nonlinear prediction and change detection techniques. The proposed methods are computationally demanding, but they can be applied to blind nonlinear problems of higher dimensions than other existing approaches.

A simple nation-wide core network architecture based on the optimized combination of WDM and OTDM technologies in a two-tier structure network is proposed. The dynamic timeslot allocation in a fixed length frame structure associated with the wavelength routing scheme creates a virtual path with variable bandwidth for edge-to-edge transport of any type of packet protocol without O-E-O conversion. The simulation results show that dynamic timeslot assignment with bandwidth reservation is the best alternative for the network bandwidth utilization efficiency. The influence of the delay caused by the physical size of the network during the request-acknowledgement process is also discussed.

There is a growing demand that mobile networks should provide quality-of-service (QoS) to mobile users since portable devices become popular and more and more applications require real-time services. Providing QoS to mobile hosts is very difficult due to mobility of hosts. The resource ReSerVation Protocol (RSVP) establishes and maintains a reservation state to ensure a given QoS level between the sender and receiver. However, RSVP is designed for fixed networks and thus it is inadequate in wireless mobile networking environments. In this paper, we propose a resource reservation protocol for mobile hosts in mobile networks. The proposed protocol extends the RSVP by introducing RSVP agents in local networks to manage the reservations. The proposed protocol reduces packet delay, bandwidth overhead, and the number of RSVP messages to maintain reservation states. We examined the performance of the proposed protocol by simulation and we got an improved performance over the existing protocols.

In this paper, we present a polynomial construction based on Jacket and Hadamard matrices over the Galois Field. The construction has two modes, one only includes matrices extension, and the other includes a center-weighted scheme for polynomial representations. Here, an "addition" scheme is used to represent matrices, which can lead to simple operations and convenient implementation of hardware.

We have developed a fiber physical contact (FPC) connector for the high-density connection of optical fibers. This connector individually aligns multiple bare fibers in micro-holes without ferrules and realizes physical contact by using the buckling force of the fibers themselves. The fiber endfaces must be tapered to allow the fibers to be inserted into the micro-holes. The endfaces must also be polished so that they realize physical contact (PC) with excellent optical performance. For each process, we examined the required shape and processing condition of the fiber endface for the FPC connector. As regards tapering, we determined the processing condition for achieving a target tapering angle and developed a non-breaking process with the optical fibers bent. In terms of polishing, we revealed that it is important for the fiber endface angle error to be less than 0.7 degrees if we are to achieve excellent optical performance. These results allowed us to fabricate an FPC connector that exhibited excellent levels of optical performance.

This paper derives a set of orthogonal polynomials for a complex random variable that is uniformly distributed in two dimensions (2D). The polynomials are used in a series expansion to approximate memoryless nonlinearities in digital QAM systems. We also study stochastic identification of nonlinearities using the orthogonal polynomials through analysis and simulations.

Since the beginning of the last two decades, many researchers have been involved in the problem of Blind Source Separation (BSS). Whilst hundreds of algorithms have been proposed to solve BSS. These algorithms are well known as Independent Component Analysis (ICA) algorithms. Nowadays, ICA algorithms have been used to deal with various applications and they are using many performance indices. This paper is dedicated to classify the different algorithms according to their applications and performances.

In the FDTD simulation of microwave circuits, a device in very small size compared with the wavelength is often handled as a lumped element, but it may still occupy more than one cell instead of a wire structure without volume routinely employed in classical extended FDTD algorithms. In this paper, two modified extended FDTD algorithms incorporating a lumped element occupying more than one cell are developed directly from the integral form of Maxwell's equations based on the assumption whether displacement current exists inside the region where a device is present. If the displacement current exists, the modified extended FDTD algorithm can be represented as a Norton equivalent current-source circuit, or otherwise as a Thevenin equivalent voltage-source circuit. These algorithms are applied in the microwave line loaded by a lumped resistor and an active antenna to illustrated the efficiency and difference of the two algorithms.

In 1999, Araki et al. proposed a convertible limited verifier signature scheme. In this letter, we propose a universal forgery attack on their scheme. We show that any one can forge a valid signature of a user UA on an arbitrary message.

In this paper, we propose the CMOS implementation of neuron models for an artificial auditory neural network. We show that when voltage is added directly to the control terminal of the basic circuit of the hardware neuron model, a change in the output firing is observed. Next, based on this circuit, a circuit that changes with time is added to the control terminal of the basic circuit of the hardware neuron model. As a result, a neuron model is constructed with ON firing, adaptation firing, and repetitive firing using CMOS. Furthermore, an improved circuit of a neuron model with OFF firing using CMOS which has been improved from the previous model is also constructed.

Recently, Stadler et al. proposed the concept of fair blind signatures to prevent the misuse of blind signature schemes from criminals. In this article, we show the proposed scheme could not meet the untraceability property of blind signature's requirements. We point out that the proposed scheme cannot provide true blind signatures.