In this paper, we discuss crosstalk equalization technique for high-speed digital transmission systems. This equalization technique makes use of the cyclostationarity of the crosstalk interferer. We first analyze the eigenstructure of the equalizer in the presence of cyclostationary crosstalk interference. It is shown that the eigenvalues of the equalizer depend upon the folded signal and interferer power spectra, and the cross power spectrum between the signal and the interferer. The expressions of the minimum mean square error (MMSE) and the excess MSE are then obtained by using the equalizer's eigenstructure. Analysis and simulation results indicate that such peculiar equalizer's eigenstructure in the presence of cyclostationary interference results in significantly different initial convergence and steady-state behaviors as compared with the stationary noise case. We also show that the performance of the equalizer varies depending on the relative clock phase of the symbol clocks used by the signal and the crosstalk interferer.

In this paper, a grey filtering approach based on GM(1,1) model is proposed. Then the grey filtering is applied to speech enhancement. The fundamental idea in the proposed grey filtering is to relate estimation error of GM(1,1) model to additive noise. The simulation results indicate that the additive noise can be estimated accurately by the proposed grey filtering approach with an appropriate scaling factor. Note that the spectral subtraction approach to speech enhancement is heavily dependent on the accuracy of statistics of additive noise and that the grey filtering is able to estimate additive noise appropriately. A magnitude spectral subtraction (MSS) approach for speech enhancement is proposed where the mechanism to determine the non-speech and speech portions is not required. Two examples are provided to justify the proposed MSS approach based on grey filtering. The simulation results show that the objective of speech enhancement has been achieved by the proposed MSS approach. Besides, the proposed MSS approach is compared with HFR-based approach in [4] and ZP approach in [5]. Simulation results indicate that in most of cases HFR-based and ZP approaches outperform the proposed MSS approach in SNRimp. However, the proposed MSS approach has better subjective listening quality than HFR-based and ZP approaches.

For tracking electromigration induced evolution of voids a diffuse interface model is applied. We assume an interconnect as two-dimensional electrically conducting via which contains initially a circular void. The diffuse interface governing equation was solved applying a finite element scheme with a robust local grid adaptation algorithm. Simulations were carried out for voids exposed to high current. An influence of the void dynamics on the resistance of interconnect is investigated. In the case of the interconnect via it was shown that a migrating void exactly follows the current flow, retaining its stability, but due to change of shape and position causes significant fluctuations in interconnect resistance.

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.

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.

In this letter, we study the performance of the iterative receiver as applied to the space division multiplexing/orthogonal frequency division multiplexing (SDM/OFDM) systems. The iterative receiver under consideration employs the soft in/soft out (SISO) decoding process, which operates iteratively in conjunction with channel estimation for performing data detection and channel estimation at the same time. As opposed to the previous studies in which the perfect channel state information is assumed, the effects of channel estimation are taken into account for evaluating the performance of the iterative receiver and it is shown that the channel estimation applied in every iteration step of the iterative receiver plays a crucial role to warrant the performance, especially at a low signal-to-noise power ratio (SNR).

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.

Appropriate test signals defined by formula or generated by algorithm are used for measuring objective QoS (Quality of Services) for voice operated telecommunication devices such as telephone and speech codec (coder-decoder). However, that for measuring residual echo characteristics in hands-free telecommunications equipped with acoustic echo canceller is under study in ITU-T Recommendation G.167. This paper describes comparative assessment of test signals for measurement of residual echo characteristics. In hands-free telecommunications, acoustical echo canceller has been developed to remove a room echo signal through the loudspeaker to the microphone in the receiving end. Performance of the echo canceller system is evaluated by residual echo characteristics expressed in echo return loss enhancement (ERLE). The ERLE can be conventionally measured by putting white noise into the echo canceller system. However, white noise is not adequate as the test signal for measuring the performance of the echo canceller, since the performance may depend on the characteristics of input test signal, and the characteristics of the white noise differ from those of real voice. Therefore, this paper discusses appropriate characteristics of real voice required for objective quality evaluation of echo canceller system. The test signals used for this verification tests were real voice (RV), white noise (WN), frequency weighted noise (FWN), artificial voice (AV), and composite source signal (CSS) depending on the approximation of real voice characteristics. As the comparative assessment results, the ERLE characteristics measured by artificial voice conforming to ITU-T Recommendation P.50 having average characteristics of real voices in time and frequency domains are almost equivalent to those of real voice and best among those test signals. It is concluded that artificial voice P.50 is satisfied with measurement of residual echo characteristics.

In a conventional file system, the directory tree is traversed to find the inode number of a file. The inode lookup performance degrades as the size of the directory tree increases. In this letter, a new directory scheme, called direct hashing directory, is proposed. The inode number of a file is the cyclic redundancy code of the file's absolute path name such that the inode number can be computed directly. The average number of disk accesses for inode lookup is 1.08, which is order of magnitude faster than the conventional directory schemes such as hashing, B tree, and sequential directory.

This paper discusses a nonlinear function for independent component analysis to process complex-valued signals in frequency-domain blind source separation. Conventionally, nonlinear functions based on the Cartesian coordinates are widely used. However, such functions have a convergence problem. In this paper, we propose a more appropriate nonlinear function that is based on the polar coordinates of a complex number. In addition, we show that the difference between the two types of functions arises from the assumed densities of independent components. Our discussion is supported by several experimental results for separating speech signals, which show that the polar type nonlinear functions behave better than the Cartesian type.

This paper presents novel techniques for blind separation and blind extraction of instantaneously mixed binary sources, which are suitable for the case with less sensors than sources. First, a solvability analysis is presented for a general case. Necessary and sufficient conditions for recoverability of all or some part of sources are derived. A new deterministic blind separation algorithm is then proposed to estimate the mixing matrix and separate all sources efficiently in the noise-free or low noise level case. Next, using the Maximum Likelihood (ML) approach for robust estimation of centers of clusters, we have extended the algorithm for high additive noise case. Moreover, a new sequential blind extraction algorithm has been developed, which enables us not only to extract the potentially separable sources but also estimate their number. The sources can be extracted in a specific order according to their dominance (strength) in the mixtures. At last, simulation results are presented to illustrate the validity and high performance of the algorithms.

In certain computer and communication systems, the significant number of byte errors are not hard errors, but a few transient bit errors confined to byte regions. This kind of byte errors are called spotty byte errors, meaning, not all, but only 2 or 3 random bits, are corrupted in a byte. Especially, the codewords of memory systems which use recent high density wide I/O data semiconductor DRAM chips are prone to this kind of spotty byte errors. This is because, the presence of strong electromagnetic waves in the environment or the bombardment of an energetic particle on a DRAM chip is highly likely to upset more than just one bit stored in that chip. Under this situation, codes capable of correcting single spotty byte errors are suitable for application in semiconductor memory systems. This paper defines a spotty byte error as a random t-bit error confined to a b-bit byte and proposes a class of codes called Single t/b-error Correcting (St/bEC) codes which are capable of correcting single spotty byte errors occurring in computer and communication systems. For the case where the chip data output is 16 bits, i.e., b=16, the S3/16EC code proposed in this paper requires only 16 check bits, that is, only one chip is required for check bits at practical information lengths such as 64, 128 and 256 bits. Furthermore, this S3/16EC code is capable of detecting more than 95% of all single 16-bit byte errors at information length 64 bits.

This paper proposes two techniques for improving the accuracy of gate-level power analysis for system-on-a-chip (SoC). (1) Creation of custom wire load models for clock nets. (2) Use of layout information (actual net capacitance and input signal transition time). The analysis time is reduced to less than one three-hundredth of the transistor-level power analysis time. Error is within 5% against a real chip, (the same level as that of the transistor-level power analysis), if technique (2) is used, and within 15% if technique (1) is used.

The goal of the maximum cut problem is to partition the vertex set of an undirected graph into two parts in order to maximize the cardinality of the set of edges cut by the partition. The maximum cut problem has many important applications including the design of VLSI circuits and communication networks. Moreover, many optimization problems can be formulated in terms of finding the maximum cut in a network or a graph. In this paper, we propose an improved Hopfield neural network algorithm for efficiently solving the maximum cut problem. A large number of instances have been simulated. The simulation results show that the proposed algorithm is much better than previous works for solving the maximum cut problem in terms of the computation time and the solution quality.

New physical models, algorithms, and parameters are needed to accurately model emerging silicon-on-insulator (SOI) devices. The modeling approaches for various emerging SOI technologies are discussed in this paper.

This paper reviews research and development on the phased array antennas (PAAs) for several applications in Japan in over past two decades. First, the author shows the historical overview of the PAA for radar, satellite and mobile communication uses. Next, this paper introduces analysis methods for the PAA. It shows mutual coupling analysis methods and pattern synthesis methods for the PAA. Furthermore, the author discusses measurement methods for the PAA. Especially, he explains the rotating-element electric-field vector (REV) method for the Japanese original PAA calibration method. Finally, the author concludes and shows future PAA technologies.

In wavelength division multiplexed networks, shared path protection provides the same level of protection against a single fiber-link failure as dedicated path protection with potentially higher network utilization. The shared path protection is more complex to provision and maintain. In this paper, we introduce a parameter, the degree of sharing, which refers to the number of protection paths that a wavelength can be assigned to on a link. We propose methods for calculating the maximum degree of sharing. We consider on-line routing and wavelength assignment (RWA) of protection paths that are established for incremental traffic using the maximum degree of sharing. Establishment of protection paths using the maximum degree of sharing can simplify the algorithm. We compare the results on the decreased calculation time with accepted connection requests for a given number of wavelengths, assuming that wavelengths are assigned according to the First-Fit policy for working paths and Last-Fit policy for protection paths. The more wavelengths are used, the more calculation time can be reduced. When the load increases, the decreasing rate of calculation time also increases.

Turbo decoding with coherent detection requires accurate channel estimation. In this paper, we consider outer-turbo channel estimation (OTCE), which carries out iterative channel estimation before turbo decoding, and inner-turbo channel estimation (ITCE), which incorporates iterative channel estimation into turbo decoding process. The average bit error rate (BER) performances with OTCE and ITCE in a frequency nonselective Rayleigh fading channel with antenna diversity reception are evaluated by means of computer simulations to be compared. It is found that although ITCE is superior to OTCE, OTCE provides the average BER performance very close to ITCE when dual antenna diversity reception is used.

This paper addresses the system throughput maximization problem for HAPS third generation cellular systems. We assume that the Stratospheric Platform is able to perform a perfect link gain estimation for all mobile terminals, such that a centralized resource allocation strategy is made possible. A classical 3G wireless scenario is considered, where traffics characterized by different bit rates coexist with Best Effort Traffic services without stringent bit rate constraints. In this scenario, we firstly envisage three Rate Assignment schemes for best effort terminals which aim at achieving the maximum system throughput subject to different bit rate constraints. For the second envisaged rate assignment scheme, which represents the best compromise between service fairness and throughput, we then propose a simplified approach that allows to noticeably decrease the implementation complexity with a slight performance degradation.

It is still an open problem to elucidate the scaling merits of an embedded SRAM with Low Operating Power (LOP) MOSFETs fabricated in 50, 70 and 100 nm CMOS technology nodes. Taking into account a realistic SRAM cell layout, we evaluated the parasitic capacitance of the bit line (BL) as well as the word line (WL) in each generation. By means of a 3-Dimensional (3D) interconnect simulator (Raphael), we focused on the scaling merit through a comparison of the simulated SRAM BL delay for each CMOS technology node. In this paper, we propose two kinds of original interconnect structure which modify ITRS (International Technology Roadmap for Semiconductors), and make it clear that the original interconnect structures with reduced gate overlap capacitance guarantee the scaling merits of SRAM cells fabricated with LOP MOSFETs in 50 and 70 nm CMOS technology nodes.