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This paper discusses selective stimulation, which is a problem common to various electrical stimulation techniques. The induction of nerve activity by electrical stimulation is a technique which is widely used in medical and health care. In order to reduce unpleasant feelings during stimulation and to diversify the induced sensations, it is necessary to control the stimulus waveform and to design the electrode in appropriate ways. From the viewpoint of nerve stimulation, the above requirement can be interpreted as a selective stimulation problem in which the target nerve fiber is stimulated and the activities of other nerves are minimized. At the present stage of electrical stimulation, however, there is no means of representing selectivity of stimulation mathematically, although precise modeling of the stimulus site is possible. Consequently, the design for selective stimulation is based on trial and error. The purpose of this paper is to construct a mathematical framework for the design of selective stimulation. The requirement for selective stimulation is formulated as a quadratic programming problem which gives a route to the optimal stimulation technique. Several simple examples are solved, and the usefulness of the proposed method is shown through comparison to selective stimulation techniques that were proposed in the past. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(9): 62–72, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20098 [ABSTRACT FROM AUTHOR]

In this paper, a formulation of the convergence property of the NLMS method, an updating algorithm for an adaptive Volterra filter, is discussed. To date, much research has been carried out on the convergence property of the updating algorithm in a linear adaptive filter. However, there has been insufficient study of the updating algorithm for the second-order adaptive Volterra filter. In this paper, the convergence property of the E-NLMS method, an updating algorithm in the second-order adaptive Volterra filter, is formulated and the convergence property of the second-order adaptive Volterra filter is theoretically determined. In the formulation of the convergence property we use the first-order recursive filter expression, which is effective for studying various characteristics of the NLMS method in linear adaptive filters. In addition, it is recognized that the nature of the input signal to the diagonal (D) elements of the second-order Volterra system is different from that of the input to the other (ED) components. By means of computer simulation, it is confirmed that the convergence property given by the derived theoretical equation can sufficiently express the actual convergence property. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(10): 40–50, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1124 [ABSTRACT FROM AUTHOR]

This paper presents a discussion of the gate in target tracking. Here, the gate is defined as the region in which the tracked target is expected to exist. The observation vector in the gate is used for tracking. For instance, in tracking in a two-dimensional plane, various shapes can be conceived as a gate, including a rectangle, circle, and ellipse. However, no publication has dealt with the shape of the optimal gate. As the gate is made larger, more clutter or observation vectors other than the tracked target are included within it, so that tracking becomes more difficult. On the other hand, if the gate is made smaller, there is an increasing danger that the observation vector from the tracked target may no longer fall within the gate. There exists no practical method of determining the gate size in terms of the target detection probability or the tracking situation, such as the clutter density. In this paper, the gate shape that maximizes the in-gate probability, indicating the probability that the tracked target falls within the gate when the gate volume is constant, is found. Further, a method for determining the in-gate probability by increasing the average rate at which the detected tracked target falls within the gate and by decreasing the average number of false signals from objects other than the target detected within the gate is presented. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(5): 10–22, 2002 [ABSTRACT FROM AUTHOR]

This paper proposes a median-type fuzzy filter, aiming at the signal estimation in the case where the impulse and white Gaussian noise are superposed on the signal with sudden changes, such as the image signal. The filter is realized by generalizing the fuzzy filter previously proposed by the authors. In the previously proposed fuzzy filter, the signal point to be estimated is taken as the reference point and the signal is estimated by determining the input signals relatively close to the reference point, using the fuzzy clustering; whereas the filter proposed in this paper uses the median value of the input signal sequence as the reference point and the signal is estimated by determining the input signals close to that reference point using the fuzzy clustering. The signal is estimated by determining the center of gravity of the cluster of the points close to the median. Furthermore, in the proposed filter, the optimum design is realized by learning with respect to appropriate training signals. It is in general not possible to decide definitely whether or not the input signal is close to the median, and thus the fuzzy clustering is effective since it takes into consideration the ambiguity of the signal. By introducing the fuzzy clustering, it is shown in particular than an effective signal estimation is realized for the signal with subtle changes. [ABSTRACT FROM AUTHOR]

Coordinated checkpointing techniques ensure that a consistent global state is maintained by means of coordination between processes. The approach requires that application messages temporarily cease to be exchanged but the rollback procedure when recovering from a fault is consequently simplified and the recovery costs are small. With current reductions in communications costs, the importance of coordinated techniques may be seen to be growing. However, in large-scale systems there is a possibility that performance will be seriously impaired due to the frequent halting of the exchange of messages. In this paper we propose a method whereby coordination is performed at only a subset of the checkpoint generation points that are periodically visited while at the remaining points each process independently generates an incremental snapshot. This method aims to both alleviate the performance degradation incurred from coordination and to realize relatively high-speed recovery. In evaluating the effectiveness of this method we estimate the checkpointing overheads and recovery costs using a probabilistic model and simulations and compare it with existing coordination methods. The results show that the proposed method is more effective than existing coordination methods from the perspective of both performance and reliability in environments with a relatively low frequency of messages. In addition, we perform comparisons of two different delta schemes for representing the incremental snapshots and discuss which environments they are each respectively suited to. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(8): 39– 53, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20296 [ABSTRACT FROM AUTHOR]

This paper shows a technique for efficiently analyzing the information flow of a program that generally includes recursive procedures. This technique enables security levels representing the secrecy of data to be assigned according to an arbitrary finite distributive lattice. This paper proves the soundness of the proposed information flow analysis method and shows that this algorithm can be executed in cubic order of the program description size. In addition, a congruence relation is used to extend the algorithm so that functions can be appropriately introduced to enable information within arguments to be hidden. Finally, a prototype system is used to present execution results of the proposed algorithm. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(9): 48–61, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20099 [ABSTRACT FROM AUTHOR]

The wavelet transform is a practical method for multiresolution analysis. Previously, methods for describing the transformed signal using zero-crossing information and additional complementing information (zero-crossing representation) have been studied. The zero-crossing representation, however, presents problems in that the physical interpretation of the applied additional information is not clear, and reconstruction of the original signal from the zero-crossing representation involves iterations of nonlinear operations. Other problems include the fact that extension to the multidimensional case is difficult and that the computational complexity is considerable. In this paper, the authors propose a new zero-crossing representation that uses the additional information with clear physical interpretation and has a form that is easy to handle in signal processing. A method is proposed to reconstruct the original signal from the zero-crossing representation. The method uses iterative linear operations with a theoretical guarantee of convergence and has lower computional complexity than the conventional method. The proposed zero-crossing representation is extended to the two-dimensional case and is applied to an actual image. It is experimentally verified that the reconstructed image is a good approximation to the original signal. Furthermore, the amount of additional information in the zero-crossing representation is reduced. The quality degradation of the reconstructed image is examined. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(1): 68–78, 1999 [ABSTRACT FROM AUTHOR]

This paper formulates the modified least-squares method presented by Mullis in a general form so that weighting can be applied. Then, all algorithm is proposed with which a least-squares approximation can be derived for an IIR digital filter, by solving iteratively a system of equations. The method is extended to a complex Chebyshev approximation that minimizes the maximum magnitude of the complex error, by multiplying the weight, which is newly derived in this paper, with the modified least-squares error. The method is compared with a conventional complex Chebyshev approximation by means of a computer simulation and the effectiveness of the proposed method is demonstrated. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 28–36, 1998 [ABSTRACT FROM AUTHOR]

In natural observation theory, input waveforms are observed using natural observation transforms. Especially in the case of the neighboring type, the effect of the uncertainty principle as it applies to a natural observation transform was clarified by using the equivalent time constant. This paper aims to characterize natural observation transforms of normal type in waveform observation from the point of view of the uncertainty principle. Besides the equivalent time constant, the average degree index is introduced to represent the uncertainty principle. It describes a relation between the equivalent time constant and the variance in the observation degree. This paper gives a new insight into the set of fundamental observed values of normal type from the point of view of observation resolution. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 45–54, 1998 [ABSTRACT FROM AUTHOR]

This paper treats multicommodity flow problems with a flow assignment constraint. General multicommodity flow problems require both the path set in which the commodities are assigned and the volume to which the commodities are assigned for each path. However. In this paper, we treat only problems wherein paths are required. The flow is assigned to the paths uniformly. This problem appears in state-and-time-dependent routing (STR), which is kind of dynamic routing scheme in telecommunication networks. We show that this problem is NP-complete and remains NP-complete if the lengths (number of edges) of paths are restricted to less than or equal to two can be solved in polynomial time if the path lengths are restricted to be less than or equal to two and the number of commodities is fixed. [ABSTRACT FROM AUTHOR]

This paper considers a plant with structured uncertainties, such as one with a transfer function where the denominator and the numerator, respectively, are represented by a convex combination of two polynomials. A necessary and sufficient condition for the robust stabilization is presented, and the systematic design method for the compensator is shown. Key words: Uncertain system; feedback control; robust stabilization; Hurwitz polynomial; interpolation. [ABSTRACT FROM AUTHOR]

This paper defines anew the maximum two-route flow as a measure to represent the relation between two vertices in the flow network. The maximum two-route flow corresponds to the maximum number of communication channels that are composed of two-route paths between two terminals in a communication network. The F-rings that can be composed simultaneously, containing the considered two vertices, is defined as the maximum two-route flow between those two vertices. It is shown in this paper that a theorem exists for the maximum two-route flow similar to the max-flow min-cut theorem for the ordinary flow. [ABSTRACT FROM AUTHOR]

In the fields of signal transmission and its processing, compensation of phase or group delay distortion is an important problem. This paper proposes an approximation method of group delay characteristic of one-dimensional (1-D) and 2-D discrete-time systems using an extended equation error function as its performance index. Since the phase approximation problem is inherently nonlinear, some iterative approaches have been proposed. Its disadvantage is in the increase of the amount of computation. Another method to transform group delay approximation problem to power spectrum approximation problem is presented. This indirect approach leads to the minimum squared error problem which is easily solved but does not yield a satisfactory result. In the first part of this paper, the 1-D group delay approximation problem is formulated to a linear problem in group delay domain. Then we apply the quasi-least mean squared method to it. In the latter part of this paper, the 1-D method is extended to the 2-D group delay approximation problem. Some examples are shown to verify the effectiveness of the method. [ABSTRACT FROM AUTHOR]

This paper proposes a design method for the FIR two-channel perfect quadrature mirror filter (QMF) with the linear phase. The two-channel perfect QMF can be designed by Vetterli's method, where a system of equations representing the condition for the perfect reconstruction of the signal is solved. The filter designed by this method, however, does not, in general, have good frequency characteristics. This paper presents the design for the two-channel QMF with a perfect reconstruction and a good amplitude characteristic. The method is based on Vetterli's method, and a constraint to approximate the amplitude charac- teristic in the frequency domain is added to the condition of perfect reconstruction in time domain. Then Remez' algorithm is applied to the derived system of equations. The construction of the two-channel perfect QMF, when the coefficient is quantized, also is discussed. Furthermore, a method is shown in which the two-dimensional (2-D) perfect QMF is designed by applying the McClellan transformation to the obtained 1-D QMF. The condition for the McClellan transformation for the perfect reconstruction is derived. [ABSTRACT FROM AUTHOR]

Fractal coding, which is coding with the self-similarity of the image as redundancy, has recently been studied extensively. In this paper, a method of fractal video coding is proposed in which the redundancy along the time axis is eliminated from a sequence of videos by the orthogonal transformation and then the results are rearranged according to the frequency components before fractal video coding. In comparison with the conventional fractal video coding, the average PSNR is improved with almost identical information content. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(3): 36–47, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20248 [ABSTRACT FROM AUTHOR]

It is known that rats with hippocampal lesions exhibit abnormal spatial navigation behavior. In order to analyze the role of the hippocampus of the rat in performing spatial navigation tasks, this paper presents a hippocampus–basal ganglia model. The hippocampus in this model integrates multiple visual information inputs from the visual association area, and the basal ganglia perform reinforcement learning in order to acquire a stochastic mapping from the information obtained from the visual association area and the hippocampus to the output action. Assuming milk maze tasks with visible and invisible platforms, numerical simulations are performed using the proposed model. The result is that both normal rats and rats with hippocampal lesions can perform the milk maze task with a visible platform. In the milk maze task with an invisible platform, on the other hand, rats with hippocampal lesions require a longer time than normal rats to arrive at the platform. These results agree with the experimental results. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(10): 66–80, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20109 [ABSTRACT FROM AUTHOR]

This paper proposes a method of generation of images taken from a virtual viewpoint, based on depth information on the object when viewed from multiple positions. First, the depth map from a viewpoint is converted to the depth map of the object as viewed from a virtual point. Then, based on the depth map from another viewpoint, the occlusion area of the virtual viewpoint depth map is complemented and the depth value is corrected. By appropriate use of real images through interpolation of depth values and smoothing, a virtual viewpoint image is generated. Applying the proposed method makes it possible to generate a virtual viewpoint image with less occlusion, even for a viewpoint deviating greatly from the actual camera position. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(10): 81–88, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20110 [ABSTRACT FROM AUTHOR]

Entropy estimation for a memory-less information source has mainly been performed by the following process under the condition that the occurrence probability of each source symbol is unknown: First, the occurrence probabilities of source symbols are estimated; then the entropy is estimated by substituting the estimated occurrence probabilities into the entropy function. Such an entropy estimation is not optimum in the sense of least squares error; it causes a large error, particularly for small sample size. In this paper, we propose an entropy estimator that strictly minimizes the mean squared error for a two-ary memory-less information source; it is formulated as a function of the number of one type of source symbols contained in the sample set. Furthermore, we propose a technique for estimating entropy of an arbitrary M-ary memory-less information source by recursively applying the proposed entropy estimator for a two-ary memory-less information source. Numerical experiments show that this technique improves mean squared error compared to the conventional method even though the guarantee that it is the least squares error estimator has been lost. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(9): 1–10, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10163 [ABSTRACT FROM AUTHOR]

In this paper, to automate broadcast monitoring, which is typified by the verification of the transmission of the material ID code of a commercial message (CM) or program, the authors propose a watermarking scheme without reference images, which does not require a reference image when the watermark sequence is extracted. By embedding a watermark sequence formed by using standard normal random numbers after discretizing the embedding-target coefficients in the DCT domain, the proposed technique makes a reference image unnecessary when the watermark sequence is extracted. In addition, since the proposed technique enables a watermark image having the desired image quality (PSNR) to be achieved for any image, it has the advantage that the image quality can always be maintained regardless of the image. Therefore, the same procedure can always be used to embed the watermark sequence in a full-motion video image without obtaining parameters for each image and to extract that watermark sequence without using a reference image. The authors used simulations to verify that an image quality of at least 56 dB, which is the HDTV digital VTR guideline, can be maintained even in the watermarked image, that the watermark sequence can be extracted without using a reference image, and that the proposed technique is robust with respect to MPEG. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(9): 22–32, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20101 [ABSTRACT FROM AUTHOR]

This paper proposes a human–robot communication system based on observations of robot–robot conversations to encourage humans to consider robots as natural targets for communication. In the proposed system, when nearby robots are communicating, voices and gestures are used to indicate that the robots are conversing with each other. When humans in the vicinity observe a robot building a relationship by interacting with another robot or the surrounding environment, they regard robots as equal targets for communication and build relationships with robots or carry out natural communications with robots. We used robots with sufficient body expression capabilities to actually implement a robot system that enabled communications according to robot–robot conversations or situations. This paper reports on a robot system that implements both of these kinds of robot–robot conversations and human–robot conversations and verifies the effects of the observation of robot–robot conversations through psychological experiments. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(5): 48–58, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10171 [ABSTRACT FROM AUTHOR]

In the optimum design of an FIR filter by the complex Chebyshev method, it is difficult to limit the increase of computational effort and to guarantee convergence to an optimum solution due to the nonlinearity of the problem and the instability of the frequency points for the optima. In this paper, the complex Chebyshev approximation problem is formulated as a linear semi-infinite programming problem by means of the real rotation theorem. An optimum design method is proposed using the three-phase method, a linear semi-infinite programming method. In the present method, the design problem is first reduced to a linear programming problem with constraints and a provisional solution is derived by the simplex method. After the constraints corresponding to the degenerated optimum frequencies contained in the provisional solution are combined, the solutions are adjusted to satisfy the optimum condition so as to obtain an optimum solution. Since low accuracy is sufficient for the provisional solution, memory requirements and computational effort can be reduced. By means of a design example, it is shown that the proposed method is superior in terms of solution accuracy and computational effort to the technique based on the simplex method. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(3): 49–57, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10104 [ABSTRACT FROM AUTHOR]

In this paper, an automatic method for the design of compact acoustic systems, using the genetic algorithm, is proposed for improvement of degraded sensitivity caused by leakage. In the present method, an analysis based on the acoustic equivalent circuit is used. The equivalent circuit and the acoustic parameters of the circuit are designed in such a way that the sound pressure frequency characteristics are contained with the range based on the GSM (ETSI EN 300 903) specifications and that the sensitivity deterioration is improved. The PfGA (Parameter-free Genetic Algorithm) is used in the design of the circuit configuration, and the genetic algorithm of Michalewicz is used for the design of acoustic parameters. As a result, an equivalent circuit is obtained that realizes characteristics not significantly affected by leakage. The present method can be applied to various compact acoustic systems such as cellular phones and provides useful information for actual design. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(3): 1–9, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10149 [ABSTRACT FROM AUTHOR]

In this paper, we propose a radial basis function–linear adaptive filter (RBF-LAF) detector that uses a radial basis function (RBF) network with an MMSE (minimum mean squared error) detector based on a linear adaptive filter (LAF) to suppress co-channel interference in DS-CDMA communications. The objective of the proposed RBF-LAF detector is to achieve an excellent bit error rate while limiting the computational complexity by calculating the received signal power or the signal-to-interference ratio (SIR) of each user and assigning an RBF detector to users receiving strong co-channel interference and an MMSE-LAF detector to users experiencing little interference. By appropriately assigning the users, an excellent bit error rate can be obtained with a relatively low computational complexity compared to conventional RBF detectors. And by using an adaptive algorithm, changes in the communication environment can be handled. Finally, the effectiveness of the proposed method is demonstrated by comparing the proposed RBF-LAF detector and the conventional detectors in terms of the bit error rate and the computational complexity in numerical examples. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(10): 19–35, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1154 [ABSTRACT FROM AUTHOR]

Clustering and labeling are basic processes in speech and image recognition. In this paper, we present the design of an analog circuit for a clustering and labeling system and the performance evaluation results based on a simulation. The number of basic computation modules becomes huge when digital LSIs are designed for parallel processing. It becomes one reason for the increase in the chip area. Therefore, we propose an architecture for a clustering/labeling circuit. It introduces the similarity circuit that utilizes the fundamental characteristics of MOS and the C matrix circuit that outputs the results of the matrix computation. We use this circuit to conduct a circuit simulation that includes statistical analysis and confirm the ability to perform feature clustering of simple speech. The proposed analog circuit has an extremely simple architecture and is expected to decrease the chip area by about 1/36 compared to an 8-bit digital design. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(10): 1–7, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1120 [ABSTRACT FROM AUTHOR]

An extremely significant transition phenomenon from a periodic phase to a chaotic phase is known as torus doubling. In this research, we perform computer simulations to study what kind of periodic behavior exists in the chaotic phase transitioned into due to torus doubling in an electrical circuit that generates torus doubling. The result was the generation of an n-fold torus that is believed to be deeply related to the period of a relatively large window that appears in the chaotic phase in logistic maps such as a 6-fold, 5-fold, or 3-fold torus in the chaotic phase. In this paper, we call the torus generated in the gap of this chaotic phase the torus window. We also analyze a coupled model consisting of a logistic map and a sine-circle map as the model to explain the mechanism for generating the torus window. We perform computer simulations to show that the 6-, 5-, and 3-fold torus windows are generated in this model. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(5): 56–62, 2002 [ABSTRACT FROM AUTHOR]

When a region with a certain area needs silencing, no sufficient effect can be obtained with a single-channel active noise control system. Therefore, multiple channel noise control systems using several microphones and speakers have been proposed. One of the algorithms used frequently is the Multiple Error Filtered-X (MEFX) method. The MEFX method requires prior estimation of the error path system. In the case of multiple channels, the estimation error poses more complex effects in comparison to the single channel. In addition, variations of the system during operation are problematic. In this paper, the simultaneous equation method recently proposed for the single channel in which no estimation is required for the error path system is applied to the multiple channel active noise control. In the present procedure, the coefficients of the noise control filter can be estimated without estimation of the error path system as in the case of the single channel. The method can handle the variation of the path during operation and can obtain excellent silencing effects. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(5): 46–55, 2002 [ABSTRACT FROM AUTHOR]

This paper proposes an error-correction algorithm for channel error using the hidden Markov model (HMM). The proposed method uses two probabilities calculated by the information source model (HMM) and the channel model, under the constraint of the received code sequence, and estimates the transmitted code sequence by maximum likelihood. Consequently, the robustness to error can be improved without adding error-correcting code by applying the proposed method to the preprocessing of the existing code system. Using the code sequence obtained by quantizing the LSP parameters, the reduction of cepstrum distortion in error correction by the proposed method is examined. It is found that the distortion can be reduced by approximately 39% by the proposed method (16-state HMM) for a code sequence with 3% random errors. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(5): 63–71, 2002 [ABSTRACT FROM AUTHOR]

A method is proposed for configuring the compact code for extended binary memoryless sources with probabilistic deviations in symbol appearance. In an n-th order extended information source for binary memoryless sources, messages with identical appearance probabilities are generated abundantly. If these messages are grouped for treatment, no rearrangement of the messages encompassing the groups is generated in degenerate operations by the Huffman algorithm for low-entropy information sources where the appearance probability of the superior symbols approaches arbitrarily close to 1. That is the situation used in this paper. It is reported that the code length and sets of code words, average code length, and maximum code length of the compact code can be derived by a simple equation without forming a code tree. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(1): 70–77, 2000 [ABSTRACT FROM AUTHOR]

In multimodal communication using visual data, a mechanism for automatically tracking the head of a user is necessary in order to provide an environment in which the user can focus on communication itself without needing to worry about the position and size of his own face or other obstacles in the visual field of the camera. In this paper, a method for such target tracking is proposed using observations based on local moments, with observation parameters regulated through α-β filters. The main features of the proposed method may be described as follows. Local moments employed as observed values are calculated for local areas of the image corresponding to the viewpoint and visual field. Due to the use of local moments, only local calculations are performed, which reduces the overall amount of computation and suppresses the influence of other objects or noise. In the problematic cases in which the target object approaches an obstacle, a mechanism is provided to recognize this fact and to avoid the obstacles. In particular, two alternative models are provided, one allowing for nearby obstacles and the other disregarding this possibility. One of the models is selected on the basis of a certain criterion, to be used for estimation of the target position. Experiments with model images and actual images confirmed the effectiveness of obstacle proximity recognition, and of the employment of alternative models, thus demonstrating that the proposed system ensures steady target tracking. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(3): 82–90, 1999 [ABSTRACT FROM AUTHOR]

In this paper, we propose a model of the human head and eyeballs that naturally simulates the movement of its line of sight in computer graphics. We first measure the actual trajectory characteristics of head movement for several angles. From previous research and the data we obtained, we find that moving the head with a sigmoidal angular velocity is an optimal feature. A mathematical moving model of the head and eyeballs based on our finding is proposed. We create several movements, such as furtive, weird, astonished, and glancing facial expressions, by changing the model parameters. We also present a simple and practical model. Finally, we discuss our proposed model based on our evaluation of subjective tests on the model. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(1): 41–50, 1999 [ABSTRACT FROM AUTHOR]

The enlargement of a digital image is a process of narrowing the sampling interval. Since the image before the enlargement and that after the enlargement have different Nyquist frequencies, it is necessary to predict or estimate the high-frequency components that are lost in the image before processing and to complement these components in the enlargement process. It is noted that the image can be represented as the sum of Gaussian components and Laplacian components in a hierarchical structure (pyramid representation). There exists a strong correlation between the Laplacian images of the layers. Based on that property, Greenspan and Anderson have proposed a nonlinear procedure where the unknown Laplacian image of high resolution (i.e., containing the lost high-frequency components) is estimated from the Laplacian image of low resolution. This paper notes that the Greenspan procedure can be network-structured and proposes an enlargement method for digital images using a neural network (NN) for improving the resolution, based on generalizing the network. It is shown that the enlarged NN can be constructed almost independently of the kind of training image or of the hierarchy (resolution). It is also shown that a better enlarged image can be derived compared with Greenspan's method. In other words, the effectiveness and practical usefulness of the proposed method are demonstrated. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(1): 19–29, 1999 [ABSTRACT FROM AUTHOR]

By the problem, called a representation of the integers, is meant the endowing of the positive integers without a known upper bound, for which unique decoding can be provided. Form the viewpoint of information theory, this problem has been discussed theoretically in its relation to the universality, asymptotic optimality, and computation theory. The result in those approaches has been applied to the data compression scheme for the type of date referring to the past as well as the URR and other floating-point number representations with variable exponential length. This paper discusses representation of the integers problem form the viewpoint of the codeword-length function, which is the function to represent the length of the codeword. As a result, a general construction method is presented which has the desirable property and can uniquely encode all representations. Several methods are given from this approach, which can construct not only the traditional representations of the integers but also the first representations of the integers which has not been considered. It is noted first that two construction methods of codeword length, which are independent of the encoding, are considered. One is based on the combination of the codeword-length functions, and the other is based on the codeword-length distribution. More precisely, it is shown that not only are the traditional representations derived, but also the representations that have not been considered in the past. Using detailed examples, it is shown that the computational complexity of the encoding and decoding depends on the codeword-length function. [ABSTRACT FROM AUTHOR]

The construction of robust communication networks is discussed from a viewpoint of graph augmentation problems. Suppose that faults in a communication network occur at terminals or links; each terminal fault disables that terminal and the incident links, and each link fault prevents communication through that link. A network with a specified set of terminals is robust with respect to k-1 link faults (k-1 terminal faults, respectively) if communication between any pair of nonfaulty specified ones continues even if k-1 link faults (k-1 terminal faults) occur. A robust network is constructed from an existing one, possibly without links, by adding a set of links. Here, a cost is assigned to each link and it represents the total cost necessary for a physical linkage between corresponding terminals. In this paper links are bidirectional and it is considered how to construct a robust network with respect to 1 link fault (1 terminal fault) by adding a set of links that minimizes the total sum of costs assigned to those links. The problem is formulated as the 2-edge connectivity (2-vertex-connectivity) augmentation problem for a specified set of vertices, and an O(|V|²) (an O(|V|³)) approximation algorithm BRA-SV (BIA-SV) is proposed. It is shown that the worst approximation of BRA-SV (BIA-SV) is no more than twice (strictly less than four times) the optimal.

In information theory, the channel capacity is defined as the maximum of the transmission rate at which information can be conveyed reliably over a noisy channel. This paper presents a new algorithm for computing the capacity of discrete memoryless channels. The problem of determining the capacity is a concave programming. Thus it can be computed by solving a system of nonlinear equations termed the Kuhn-Tucker equations. Newton's method often is used to solve nonlinear equations. However, it does not have global convergence and may not converge when the starting point is not close to the solution. A globally convergent algorithm using simplicial subdivision has been proposed for the capacity evaluation, but it is very slow and cannot be applied to large-scale problems. In this paper, the Kuhn-Tucker equations are transformed into a system of equations with separable mappings by introducing auxiliary variables. This system of equations can be solved by a homotopy method using rectangular subdivision, which Is much more efficient than the simplicial-type algorithm. Moreover, the proposed algorithm has quadratic convergence, so it converges rapidly similarly to Newton's method, The global convergence of the proposed algorithm is also proved herein. By numerical examples, it is shown that the proposed algorithm is more efficient than the welt-known Arimoto algorithm.

The two ways to decode the error-correcting code are by hard-decision decoding and soft-decision decoding. Soft-decision decoding can improve the code gain by 2 to 3 dB, compared with hard-decision coding, but the decoding system becomes complicated. Consequently, it has not been applied widely except in some special cases such as extraterrestrial communication. However, with the remarkable progress in LSI technology as the background, its application is now being considered for terrestrial communication and recording technologies. The error-correcting code used in the application has also changed from the code which requires only a simple decoding system, such as Hamming code, to the Reed-Solomon (RS) code, which has a higher error-correcting ability. In the soft decision for a multilevel code such as RS code, the generation of the analog weight, which indicates the reliability of the received symbol, is a problem. Forney has presented a generation method for the analog weight for the multilevel symbol in CMD decoding, which is a soft-decision decoding with a high efficiency (called optimum analog weight generation). However, the method has a problem in that the generation requires a tremendous amount of computation, and it is practically impossible to apply the GMD decoding to the multilevel code. By contrast, this paper considers a 2n code, and proposes two practical symbol analog weight-generation methods, which require only 4n/5X2n and n/5X2n computations, respectively, for the generation of the optimum analog weight. By this scheme, GMD decoding is made practically applicable to the 2n code. It is shown by numerical computation and computer simulation that the forementioned reduction of computation does not lead to the degradation of the GMD decoding efficiency. [ABSTRACT FROM AUTHOR]

Some methods for synthesizing a protocol specification from a given service specification have been proposed using various computational models. However, the existing methods cannot treat service specifications including both complex control flows and system variables. In this paper, we propose a method for synthesizing a protocol specification automatically from a pair consisting of a service specification and a resource allocation. In our method, service specifications are described in an extended model of Petri nets, which can treat system variables. We have developed a pair consisting of a synthesis system and an execution system, and applied them to CSCW (Computer Supported Cooperative Work). © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 81(8): 18–26, 1998 [ABSTRACT FROM AUTHOR]

This paper describes a new HMM synthesis method in which HMM adapts to additive noise and multiplicative distortion. The conventional HMM synthesis method can only be applied to additive noise. In the method described here, the likelihood of the synthesized HMM to adapt to speech is maximized, so that multiplicative distortion is eliminated when the method is applied. Within the framework of this method, adaptation to variations in the SN ratio, considered a problem in conventional HMM synthesis, can be formulated as part of the adaptation to multiplicative distortion. As a result of evaluating speech recognition rates using our method, we have confirmed that the method is effective for improving the recognition rate of speech that contains additive noise and multiplicative distortion. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 81(8): 1–9, 1998 [ABSTRACT FROM AUTHOR]

A visual secret sharing scheme [1] permits a secret to be shared among participants using transparencies. In this paper, we consider an extended scheme for visual secret sharing. We propose an improved method of constructing the visual secret sharing scheme that can conceal some images in a series of transparencies, in such a way that different images are seen as the number of stacking transparencies increases. Furthermore, we describe applications of the visual secret sharing scheme to copy machines and human identification schemes. In the identification scheme, users recognize messages from an identification terminal by stacking their transparencies on the display. A great advantage of this scheme is that users can validate the authenticity of the terminal without consulting a computer or calculator. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 55–63, 1998 [ABSTRACT FROM AUTHOR]

The general quantum communication process, with the optical communication process as its actual model, is strictly formulated mathematically in terms of Ohya's quantum mechanics channel, which describes the state change. Until recently the decay process and the amplifier process have been described as actual communication processes, using the concepts of the channel and its extension, lifting. Based on that description, the error probability and SNP are derived. This paper introduces a new system called the control system into the communication process, and provides the squeezed vacuum state as the quantum state of the system. We examine how conventional communication efficiency can be improved by the squeezed vacuum state, using the error probability. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 18–27, 1998 [ABSTRACT FROM AUTHOR]

In this paper, an adaptive array is proposed in which the CMA (Constant Modulus Algorithm) and LMS algorithms are combined so that a pass diversity effect is obtained without using training signals. The basic operations of the proposed algorithm are as follows. CMA captures the direct wave with maximum power. Based on the output of the CMA adaptive filter, a reference signal is generated. The LMS algorithm, using this reference signal, captures the delayed wave. By combining the direct wave and the delayed wave at maximum rate, the receiving quality is improved. By means of a computer simulation, we confirmed that the convergence property is improved and that the effect of the reference signal error on the system convergence characteristics is small. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 37–44, 1998 [ABSTRACT FROM AUTHOR]

Recently, many methods have been reported for transforming fuzzy multiple-objective linear programming (MOLP) problems, However, the transformation into a crisp MOLP problem requires an extremely large number of constraints and decision variables which is the main disadvantage of these approaches. This paper proposes that a maximizing MOLP problem with trapezoidal positive fuzzy coefficients can be transformed into a crisp MOLP problem without increasing the number of constraints by basing the two indices on the inequality relation between trapezoidal fuzzy numbers. The usefulness of the proposed method is demonstrated. [ABSTRACT FROM AUTHOR]

This paper considers the controlled discrete event systems modeled by Petri nets with external input places. A maximal element in the set of permissive feedbacks is called a maximally permissive feedback. First, a simpler condition is derived for its unique existence which does not require constructing the set of permissive feedbacks. Next, a necessary and sufficient condition is derived for the reachable set under the unique maximally permissive feedback to be equal to the set of markings which satisfy the given predicate. Moreover, synthesis of the unique maximally permissive feedback is demonstrated for a simple manufacturing system. [ABSTRACT FROM AUTHOR]

The circuit with zero sensitivity at the matching points is desirable in the sense that a very low sensitivity can be expected in the passband. It should also be noted that the wave digital filter (WDF) with a II-cascade structure has the advantage that the sensitivity in the stopband is lower than WDF with a T-cascade structure. However, the synthesis method has not been known for WDF with a II-cascade structure with the zero amplitude sensitivity at the matching points which can realize an arbitrary transfer function. This paper is an attempt to realize such a WDF. First, it is shown what type of sections should be used as the reference section of a doubly resistor-terminated reactance filter. The two terminating resistances are set equal to avoid the situation where the sensitivity at the matching point turns out to be nonzero because of the effect of the terminating resistance ratio. Then the replacement of the multiplier is shown so that the forementioned state is maintained, even if the multiplier coefficients are quantized. It is shown that WDF exists corresponding to each section, where the forementioned replacement can be applied and the structural boundedness is retained even after the coefficients are quantized. As a result, it is shown that WDF actually exists with a II-cascade structure with the zero amplitude sensitivity at the matching points which can realize an arbitrary transfer function. [ABSTRACT FROM AUTHOR]

A derivation of the Riccati equation giving a necessary and sufficient condition for H ∞ control by state feedback is presented. It is proved by solving the LQ-max problem which is equivalent to the satisfaction condition (H ∞ norm constraint condition) such that the uniform norm ∣Tvz∣∞ of a transfer operator T from the disturbance v to the controlled output z is kept below a prescribed γ. Based on the state space representation, the proof is achieved by extending optimal regulator theory. While normal proofs are based fundamentally on the analysis of the H ∞ norm in the frequency domain, it is performed fully in the time domain. This paper provides a direct and self-contained proof. [ABSTRACT FROM AUTHOR]

A length extensional mode quartz crystal resonator of small size and good antishock performance and which functions at relatively low frequencies of 200 kHz to 1.536 MHz has been used in pagers and IC cards. An on/off repetition in quartz resonators of this type is common and therefore the start-up transient performance becomes important. This paper describes a length extensional mode quartz crystal resonator with plural vibration bars which has a short start-up time. The objective of this paper is to clarify both theoretically and experimentally motional inductance L1, frequency characteristics and temperature dependence of resonance frequency, and also electric properties of the resonator. Motional inductance L1 is derived first. Especially, L1 is found theoretically to be inversely proportional to the number of bars. Next, a vibration analysis of bars and supporters is made by taking into account the lateral motions based on an energy method and then a frequency equation is derived. The frequency characteristics, temperature dependence of resonance frequency, and electric properties are analyzed and compared with experimental results. The calculated results are found to be in agreement with the experimental ones. Finally, it is shown that a length extensional mode quartz resonator exhibits a short start-up time, since L1 and R1 are compared to the single-bar case. Such resonators are believed to be manufactured in a small form and also exhibit a good anti-shock performance since vibrational and supporting portions can be made as one solid body by an etching method. Keywords: Length extensional mode quartz crystal resonator; plural vibration bars; motional inductance; start-up time; frequency temperature behavior [ABSTRACT FROM AUTHOR]

In this paper, the authors propose a new noise suppression technique that executes iterative processing and sets parameters that are suited for that processing in the spectral subtraction method, which is a noise reduction technique for noise-added speech. Iterative processing is a technique in which speech enhancement processing is executed by considering the estimated speech that is obtained when noise reduction processing is executed once as the input signal again so that a reduction of the residual noise is anticipated. A further reduction of residual noise for which speech degradation is controlled can be achieved by adjusting the parameters for each iteration. The authors also simultaneously propose a technique for maintaining the real-time nature of spectral subtraction when the proposed technique is executed. They use actual speech to which white noise, automobile noise, and crowd babble noise is added to compare the characteristics of the two proposed methods with the conventional spectral subtraction method and its improvements. The authors verified according to objective and subjective evaluations that each proposed technique showed superior results in all noisy environments. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(4): 39 –51, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20242 [ABSTRACT FROM AUTHOR]

In this paper, we propose a new adaptive filter and adaptive nonlinear predictor to implement the nonlinear prediction of speech signals to solve the problem of impulse-shaped residual errors in the linear prediction. While using order statistics processing, we add an improvement that keeps time information about the input signal to the LMS-L filter proposed by Pitas and Venetsanopoulos. Tests using synthesized speech show that the proposed nonlinear predictor has superior prediction performance to the Volterra series predictor and not just the linear predictor. However, even with this proposed nonlinear predictor, the prediction accuracy degrades in experiments using real speech. Therefore, we propose a new iterative method to conquer this problem. The iterative method pays attention to the periodicity of the speech, reuses the speech data, and obtains a prediction accuracy similar to when using many data samples. In experiments on real continuous speech, the proposed predictor improved prediction accuracy by using the iterative method, and the effectiveness of the method was confirmed. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(11): 28–42, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20148 [ABSTRACT FROM AUTHOR]

Knowledge and experience that a viewing subject possesses about an object plays a significant role in human visual recognition. In this paper we conduct psychological experiments to investigate how a subject's recognition of objects changes as the subject acquires knowledge of the category to which it belongs. In these experiments we investigated how a subject's similarity judgments with respect to different typical stimuli within a given category varied before and after category knowledge was acquired. The results of these experiments show that in many cases the acquisition of category knowledge has the effect of increasing the similarity between different patterns and in particular we are able to conclude that this effect is strong for objects that are highly typical of a category. From these results we suggest that category knowledge works to emphasize features that are strongly tied to the category knowledge. In addition, we used a computational model to conduct a simulation of the process by which category knowledge emphasizes features in a top-down manner and were able to accurately re-create the results of the psychological experiments. From these results we suggest that in human visual recognition, category knowledge plays a functional role by which features of a given object are inferred in a top-down manner from the given knowledge. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(11): 43–55, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20200 [ABSTRACT FROM AUTHOR]

In this paper, the authors propose a technique for generating creative conversations between users and multiple agents to support knowledge creation in a community. First, they describe self-expressing alter-ego agents as user proxies. An alter-ego agent accumulates statements that were originated by the user in past conversations as self-expressions and converses with other alter-ego agents by outputting these again in a manner adapted to the circumstances. Next, the authors describe the EgoChat system, which is an environment for conversations between alter-ego agents and users, and a framework for supporting knowledge creation by using EgoChat. Finally, they examine and discuss methods of evaluating this technique based on conversation examples using EgoChat. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(11): 56–65, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20201 [ABSTRACT FROM AUTHOR]

With the progressive decreases in size and increases in the speed of VLSI, guaranteeing signal integrity has become important. This paper focuses on crosstalk as a factor responsible for the violation of signal integrity and proposes a method of testing for crosstalk faults. The built-in self-test (BIST) method, which can detect crosstalk faults by using an M-sequence of a linear feedback shift register (LFSR), has been proposed. But since the output of the LFSR is used unaltered as a random test pattern, an insufficient number of crosstalk pulses is excited and some faults escape detection. We therefore propose the use of a new testing circuit which increases the number of crosstalk pulses that are excited and makes possible the detection of faults that were previously undetectable. The new testing circuit uses test points and phase shifter circuits which can output sequences that have arbitrary delays from the M-sequence. In addition, by changing the simulation method, it also becomes applicable to large-scale circuits for which such simulations could not be used in the past. We evaluate the detection rate and area overhead for several benchmark circuits and show that the method is effective in detecting crosstalk faults. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(10): 54–65, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20108 [ABSTRACT FROM AUTHOR]