This paper addresses the problem of estimating the parameters of multivariate ARMA processes by using higher-order statistics called cumulants. The main objective in this paper is to extend the idea of the q-slice algorithm in univariate ARMA processes to multivariate ARMA processes. It is shown for a multivariate ARMA process that the MA coefficient matrices can be estimated up to postmultiplication of a permutation matrix by using the third-order cumulants and of an extended permutation matrix by using the fourth-order cumulants. Simulation examples are included to demonstrate the effectiveness of the proposed method.

We present distributed load balancing mechanisms implemented on multiprocessor systems for real time video encoding, which dynamically equalize load amounts among PE's to cope with extensive computing requirements. The loosely coupled multiprocessor system, e.g. a torus connected one, is treated as the objective system. Two decentralized controlled load balancicg algorithms are proposed, and mathematical analyses are provided to obtain some insights of our decentralized controlled mechanisms. We also prove the proposed algorithms are steady and effective theoretically and experimentally.

In the band-limited signal space, the mutual relation between continuous time signal and discrete time signal is expressed by the sampling theorem of Whittaker-Someya-Shannon. This theorem consists of an orthonormal expansion formula using sinc functions. In that formula, the expansion coefficients are identical to the sample values of signals. In general, the bandlimited signal space is not always suited to model the signals in nature. The authors have proposed a new model for signal processing based on finite times continuously differentiable functions. This paper aims to complete the sampling theorem for the spline signal spaces, which corresponds to the sampling theorem of Whittaker-Someya-Shannon in the band-limited signal space. Since the obtained sampling theorem gives the simplest representation of signals, it is considered to be the most fundamental characterization of spline functions used for signal processing. The biorthonormal basis derived in this paper is considered to be a system of delta functions at sampling points with some continuous differentiability.

Wave digital filters are a class of digital filters. They are equivalent to commensurate transmission line circuits synthesized with uniform, lossless, and commensurated transmission lines. In order to extend their applications to physical wave phenomena including quantum electronics, it is necessary to consider a generalized distributed line whose velocity of energy flow has frequency characteristics. This paper discusses a generalized distributed circuit, and we obtain two types of lines, lossless and cut-off. In order to analyze these lines, we discuss signal flow graphs of steady state voltage and current. The reflection factors we obtain here are the same as that for an active power or a diagonal element of a scattering matrix, which is zero in conjugate matching. By using this reflection factor, we obtain band-pass filters synthesized with the cut-off lines. We also describe an analysis method for nonuniform line related to Riccati differential equation.

It is an important problem whether or not we can reject the disturbances from distributed parameter circuit. In order to analyze this problem structurally, it is necessary to investigate the basic equation of distributed parameter circuit in the framework of state space. Since the basic equation has two parameters for time and space, the state value belongs to an infinite-dimensional space. In this paper, the disturbance-rejection problems with incomplete state feedback and/or incomplete state feedback and feedforward for infinite-dimensional systems are studied in the framework of geometric approach. And under certain assumptions, necessary and/or sufficient conditions for these problems to be solvable are proved.

This paper introduces some modelling methods of time-varying stochastic process and its linear/nonlinear adaptive identification. Time-varying models are often identified by using a least square criterion. However the criterion should assume a time invariant stochastic model and infinite observed data. In order to adjust these serious different assumptions, some windowing techniques are introduced. Although the windows are usually applied to a batch processing of parameter estimates, all adaptive methods should also consider them at difference point of view. In this paper, two typical windowing techniques are explained into adaptive processing. In addition to the use of windows, time-varying stochastic ARMA models are built with these criterions and windows. By using these criterions and models, this paper explains nonlinear parameter estimation and the property of estimation convergence. On these discussions, some approaches are introduced, i.e., sophisticated stochastic modelling and multi-rate processing.

This paper presents techniques for generating the input patterns for locating logic design errors (PLE's) by Boolean function manipulation based on binary decision diagrams (BDD's). One PLE has one Boolean variable X or and constant values. A primary output of a correct circuit takes value X, while the designed circuit takes either 0 or 1. By using PLE's, the X-algorithms locate single or multiple logic design errors in a combinational circuit. Although PLE's play the most important role in the X-algorithms, the condition under which PLE's exist has not been formalized. This paper gives a formal analysis on the existence condition of PLE's. It is shown that the condition is always satisfied by incorporating another type of PLE. From the condition, an implicit representation of PLE's is derived. In addition, two kinds of approaches are presented for generating PLE's by Boolean function manipulation based on BDD's. One is an approach for generating all the existing PLE's. The other is a heuristic approach to obtain a limited number of PLE's in a short time. Both approaches generate PLE's including don't cares. Incorporating them, a compact representation of PLE is achieved. Experimental results have shown the compactness of the proposed representations and the availability of the pattern generation techniques.

Fluency approach is to deal with staircase, polygonal and band-limited signals as those in a unified series of signal spaces of which characteristics vary with the parameter of degree. Scaling functions and their duals have been obtained which fulfill a part of the requirements to constitute a multiresolutional analysis in this approach. The purpose of the present paper is to derive general formulae to express wavelets and their duals which fulfill the rest of the requirements. It is the first step to have a general expression of every possible wavelet in selecting a wavelet. The degree is limited to be arbitrary positive oddintegers so far in this paper. The genaral formulae derived in this paper are in the form of linear combinations of the sampling functions, which are scaling functions, and their duals. These formulae can be also regarded as a reduced version of the conditions for multiresolutional analysis in terms of sampling functions and their duals. The general formulae provide a start point for selecting a wavelet which decides characteristics of a multiresolutional analysis in the fluency approach. Some criteria for the concrete selection for each purpose of multiresolutional analysis and a formula for the even-degree cases are yet to be aquired in the future.

Relating to the problem of suppressing the immanent redundancy contained in an image with out vitiating the quality of the resultant approximation, the interpolation of multi-dimensional signal is widely discussed. The minimization of the approximation error is one of the important problems in this field. In this paper, we establish the optimum interpolatory approximation of multi-dimensional orthogonal expansions. The proposed approximation is superior, in some sense, to all the linear and the nonlinear approximations using a wide class of measures of error and the same generalized moments of these signals. Further, in the fields of information processing, we sometimes consider the orthonormal development of an image each coefficient of which represents the principal featurr of the image. The selection of the orthonormal bases becomes important in this problem. The Fisher's criterion is a powerful tool for this class of problems called declustering. In this paper, we will make some remarks to the problem of optimizing the Fisher's criterion under the condition that the quality of the approximation is maintained.

We discuss estimation error as a basic problem in formant estimation in the analysis of speech of very short-time duration in the glottal closure of the vowel. We also show in our simulation that good estimation of the first formant is almost impossible with the ordinary method using a waveform cutting. We propose a new method in which the cut waveform, as a discontinuous function of finite time, is mapped to a continuous function defined in the whole time domain; and we show that using this method, the estimation accuracy for low frequency formants can be greatly improved.

The generation and design of a stationary Markov signal are discussed as an inverse problem, in which one looks for a transition probability when a stationary probability distribution is given. This paper presents a new solution to the inverse problem, which makes it possible to design and generate a Markov random signal with arbitrary probability distribution and an exponential correlation function. Several computer results are illustrated in figures.

This paper reviews recent progress in adaptive signal processing techniques for digital mobile radio communications. In Radio Signal Processing (RSP) , digital signal processing is becoming more important because it makes it relatively easy to develop sophisticated adaptive processing techniques, Adaptive signal processing is especially important for carrier signal processing in RSP. Its main objective is to realize optimal or near-optimal radio signal transmission. Application environments of adaptive signal processing in mobile radio are clarified. Adaptive equalization is discussed in detail with the focus on adaptive MLSE based on the blind algorithm. Demodulation performance examples obtained by simulations and experiments are introduced, which demonstrates the recent advances in this field. Next, new trends in adaptive array processing, interference cancelling, and orthogonalization processing are reviewed. Finally, the three automatic calibration techniques that are based on adaptive signal processing are described for realizing high precision transmission devices.

A receiving system suitable for multipath fading channels with co-channel interference is described. This system is equipped with both an M-sectored directional antenna and an adaptive equalizer to mitigate the influence due to multipath propagation and co-channel interference. By using directional antennas, this receiving system can separate desirable signals from undesirable signals, such as multipath signals with longer delay time and co-channel interference. It accepts multipath signals which can be equalized by maximum likelihood sequence estimation, and rejects both multipath signals with longer delay time and co-channel interference. Based on computer simulation results, the performance of the proposed receiving system is analyzed assuming simple propagation models with Rayleigh-distributed multipath signals and co-channel interference.

This paper proposes a digitalized quadrature modulator for burst-by-burst carrier frequency hopping in TDMA-TDD systems. It employs digital frequency synthesis and a multiplexing modulation scheme to give the frequency offset to the modulated IF signal. Moreover, to reduce the frequency settling time of the RF synthesizer below the guard time duration, a phase and frequency preset (PFP) PLL synthesizer is employed. By employing the digital modulation scheme, the proposed modulator needs only one D/A converter, as a result, the complexity of adjusting the DC offset and amplitude between analog signals of the in-phase and the quadrature phase is eliminated. The performance of the proposed modulator is analyzed theoretically and simulated by computers. Theoretical analyses show that the frequency settling time with 15MHz hopping width in the 1900MHz band is reduced by more than 75% from that of the conventional synthesizer. The settling time is less than 40µs which is shorter than the typical guard time of the burst signal format. The analyses also show that the power consumption of the proposed modulator is lower than that of the conventional modulator employing a full band digital frequency converter. Furthermore, the computer simulation confirms that the power spectra and the constellations of the proposed modulator for the coherent and the π/4-shift QPSK modulation schemes can be successfully generated.

The blocking probabilities of n64Kb/s multi-slot calls are generally much higher than that of single slot calls. In order to improve these blocking probabilities of multi-slot calls, we propose a scheme to limit the number of time slots to be searched for lower rate calls. We analyze the performance of our scheme in a double-buffered time-space-time switching network which accommodates multi-slot calls as well as single-slot calls. The proposed method yields the reduced blocking probabilities of multi-slot calls, the increased traffic handling capacity and the reduced CPU processing load, compared with those of the conventional methods.

This paper describes the performance analysis of a distributed antenna system which includes space and path diversity with radio channel estimation. This system is used for CDMA personal communication systems. In this paper, the performance of a diversity system is analyzed precisely considering multipath and inter-antenna interference. In a diversity system, the adaptive RAKE receiver which estimates the characteristics of a radio channel adaptively has been used for diversity combining. In the adaptive RAKE, the time-variant characteristic has been approximated by a time function. In this paper, the estimation performance of the adaptive RAKE is analyzed in cases of time functions of 0-th, first and second degrees. The performances are evaluated and compared with the differential RAKE. The adaptive RAKE is found to improve the signal quality of more than 2dB in comparison with the differential RAKE. It is also found that the optimum parameter design can be achieved flexibly for radio channel estimation by using higher degree time functions.

We fabricated junctions with a porphyrin polyimide (PORPI) monolayer, and then investigated the electron transport properties of the junctions from the current-voltage (I-V) and d2V/dI2-V measurements. Polyimide LB films without porphyrin were used as tunneling barriers. One large peak was seen at a voltage around 1.9 V, due to the excitation of electron transitions in PORPI molecules, whereas a step structure was not observed in the I-V characteristic.

Reduction of the complexity of the NLMS algorithm has received attention in the area of adaptive filtering. A processing cost reduction method, in which the component of the weight vector is updated when the absolute value of the sample is greater than or equal to the average of the absolute values of the input samples, has been proposed. The convergence analysis of the processing cost reduction method has been derived from a low-pass filter expression. However, in this analysis the effect of the weignt vector components whose adaptations are skipped is not considered in terms of the direction of the gradient estimation vector. In this paper, we use an arbitrary value instead of the average of the absolute values of the input samples as a threshold level, and we derive the convergence characteristics of the processing cost reduction method with arbitrary threshold level for zero-mean white Gaussian samples. From the analytical results, it is shown that the range of the gain constant to insure convergence and the misadjustment are independent of the threshold level. Moreover, it is shown that the convergence rate is a function of the threshold level as well as the gain constant. When the gain constant is small, the processing cost is reduced by using a large threshold level without a large degradation of the convergence rate.

Electrical and optical properties of organic multilayer structure have been investigated. Two types of current-voltage characteristics have been found for thin multilayer structure of organic films. Optical property and its application for electroluminescent diode have been presented. The diode characteristics have been discussed in terms of energy band scheme.

The optical waveguides containing phthalocyanine as an optically active material were fabricated and transmission properties were investigated experimentally and numerically. The positive refractive index change was observed in the glass waveguide with a vanadyl phthalocyanine thin film as a top layer. The thermal influence on refractive index change was estimated by surface plasmon measurements.