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In state-space systems, since the minimum statistical sensitivity structures obtained by the equivalent transformations are generally completely dense, they have defects and the numbers of parameters are very large. If this point can be improved, realizations with lower sensitivity and fewer parameters are obtained. This paper presents a synthesis of new minimum statistical sensitivity structures with fewer parameters as one such realization. First, to compare sensitivities of various realizations, assuming that only noninteger elements in the coefficient matrices contain small variations, the statistical sensitivity is defined. Since the sensitivity is invariant under the orthogonal transformations, using the symmetry of the coefficient matrices of the balanced realizations which are the members of the minimum statistical sensitivity structures, it is shown that a minimum statistical sensitivity structure can be realized with fewer parameters by defining an appropriate orthogonal transformation matrix. The number of parameters required in the proposed structure depends on the pole-zero configuration of the transfer function. Finally, various realizations are compared using numerical examples. As the parameters decrease, the insensitive elements increase. Therefore, the realization proposed in this paper has the lowest sensitivity. [ABSTRACT FROM AUTHOR]

This paper considers the determination of the frequency of a sinusoidal signal from its sampled values. A method is described by which the effect of the noise is reduced and the frequency is determined accurately. The method forms the sample set composed of four samples from the signal with unknown frequency, and the binary representation of the estimation is determined successively by varying the sampling frequency. However, the noise in the signal may produce an error. One means to eliminate the effect of the noise is to determine one bit of the binary number based on several sample sets. This paper proposes the following method to reduce the estimation error for the frequency due to the noise. The binary number is partitioned into the upper and lower bits. The upper bits are determined by the property of the rum, and the lower bits are determined by the property of the ramdom-walk. In either case, each bit is determined by several sample sets. By the evaluation based on the bit error rate, the accuracy is improved by combining the forementiomed methods, compared with the majority decision. A comprehensive evaluation was also made by simulation, indicating that the r.m.s. error can be reduced to 2 percent for the signal-to-noise ratio of 20 dB. [ABSTRACT FROM AUTHOR]

This paper presents the new upper bounds of channel width required in multi- layer channel routing under a wiring rule in which horizontal wires must be wired in the odd-numbered layers and vertical wires in the even-numbered layers. In (M ≧ 4)-layer channel routing, it is proved the upper bound is equal to [(dc +k)/[M/2]], where dc is the channel density, and k is 9 when M = 5, 7 when M is even, and 13 when M is odd and greater than 5. These upper bounds are obtained by presenting the multiple-layer channel routing algorithms, which wire the channels in a column-by-column manner. [ABSTRACT FROM AUTHOR]

In a workflow information processing system, usually optimization of the workflow overall and design of each process which comprises the workflow are performed separately. As a result, inefficient procedures and duplication of processes remain in the workflow. In order to resolve this problem, the authors apply activity-based costing management (ABM), a managerial accounting method, to workflow design in this paper. Note, however, that under a conventional ABM which allocates cost and time, sufficient analytical results cannot be obtained. Thus, in this paper the authors propose a new analysis method which allocates data items in the system. Under this method, a data item in the system or an operator process is defined as “information,” and then classified as value-added information or non-value-added information depending on whether or not it contributes to the manufacturing resource produced by the system. The authors apply their proposed method to the workflow of the research paper manuscript system at the Institute of Electronics, Information, and Communication Engineers and confirm that it reduces the number of steps in the workflow by 12%. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(8): 55–65, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20087 [ABSTRACT FROM AUTHOR]

Earlier, the authors attempted an explicit evaluation of the total variation of best piecewise polynomial approximations and obtained a result [9]. However, there, the discussion was on polynomials and cannot be applied to other families of functions. Thus, the present paper considers a similar discussion with respect to the trigonometric functions, that is, the Fourier series expansion. Specifically, the total variation of Fourier series approximations is investigated, and, in addition, the norms of Fourier coefficients are evaluated. For the norms of Fourier coefficients, we confirm that they converge to zero when the degree m → ∞ and its rate is O(m-1) (Riemann-Lebesgue lemma). Conventionally, the norm of Fourier coefficients had been evaluated mostly by assuming the periodicity of functions to be approximated. However, in this paper, we evaluate the norm without using the periodicity condition of the approximated functions. Moreover, it is shown that the upper bound of the ratio between the total variations of K-th partial sum of Fourier series and that of the function to be approximated, is proportional to K. This implies that compared to the evaluation of total variation of the best piecewise polynomial approximations obtained in the authors' previous paper, this evaluation excels it by far. [ABSTRACT FROM AUTHOR]

A variational Bayes method is proposed as an approximation method for implementing Bayesian posterior distribution with moderate computational complexity. Its effectiveness for real problems has been confirmed. The variational Bayes method is a method which generalizes the mean field approximation used in calculating partition functions in statistical physics. In recent years, the mathematical properties of the precision with which it is approximated have been investigated. In this paper, the authors consider the asymptotic stochastic complexity in the case of applying the mean field approximation to complete bipartite graph-type Boltzmann machines and theoretically derive that asymptotic form. Also, based on the results, the authors quantitatively consider the difference between Bayesian posterior distribution and the posterior distribution of the mean field approximation. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(9): 1– 9, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20307 [ABSTRACT FROM AUTHOR]

The reference signals used in a multichannel system are usually strongly crosscorrelated. Preprocessing is normally added to reduce the crosscorrelation between the reference signals because this correlation makes estimating the adaptive filter coefficients difficult. However, this preprocessing is equivalent to warping the reference signals. The transmission of these warped reference signals to an unknown system hinders the essential system operation. On the other hand, it is necessary for uncorrelated components between the reference signals to exist when estimating the adaptive filter coefficients. Preprocessing that increases the ratio of these components is important in improving the convergence characteristics. In this paper, we propose an algorithm that does not transmit preprocessed reference signals to an unknown system, but only uses them to estimate the adaptive filter coefficients, and explain the design conditions needed to implement this algorithm. In other words, with the condition of not increasing the estimation errors, we derive the optimum values of the coefficients that reduce the correlation between the reference signals, which is a feature of this algorithm, and verify its effectiveness in simulations. By employing the proposed algorithm, improvements in the convergence characteristics can be designed without affecting the essential operation of the system. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(3): 32–41, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20092 [ABSTRACT FROM AUTHOR]

This paper considers the recognition of moving images and proposes a new framework in which the local features, the global structure, and motion information are handled comprehensively. The method is applied to the extraction of facial expressions and its effectiveness is demonstrated. In most conventional moving image recognition methods, recognition is performed on the basis of the sequence of recognition results for individual frames. However, the authors believe that the motion information should be more positively utilized in the recognition of the individual frames. When the time-series data are combined into the recognition of a still image, however, the amount of information is tremendous, making the processing complicated. The method proposed in this paper extends the concept of the labeled graph matching method, which has been used for still images, to moving images. The proposed method handles sparse graphs and can prevent an increase in the amount of computation. By dynamically adjusting the features to be handled according to the stage of processing, complex processing in dynamic image recognition can be integrated in a simple and straightforward way. As practical examples, the human head, and parts of the head, are extracted, indicating the effectiveness of the proposed method. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(10): 35–43, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20106 [ABSTRACT FROM AUTHOR]

There are essentially two approaches in the Ziv- Lempel code to be used in the data compression. They have already been proposed also as the improvements of the Ziv-Lempel code, which can be considered as combinations of the two methods. This paper shows that by executing the combination, not only are advantages produced, but also the structure of the Ziv-Lempel code is derived which realizes a more efficient coding. A precise data compression method is proposed based on the derived structure. The proposed method corresponds to an improvement of the LZ77 method. As the pointer information for the substrings for the coding the parameter corresponding to the length of the substring is encoded first, in contrast to most of the past methods. Then, utilizing the result the coding method for the position of the reference substring is adjusted selectively. The proposed method is one method with the highest compression ability among the modifications of the Ziv- Lempel code. [ABSTRACT FROM AUTHOR]

This paper introduces a feedforward network with a new learning algorithm into which both a self-organized mechanism and a supervised training mechanism are implemented. In particular, from some experiments, it is seen that this network can realize sophisticated signal processing and analysis data with high-speed training ability. The network is divided into two layers, that is, a self-organized layer and a supervised layer. However, in the self-organized layer, the training data of the network also are used for realizing high precision. In addition, the supervised layer also is trained in order to realize quite high accuracy for data recognition. In other words, the training data are used for both layers. Accordingly, the total precision of the network is improved greatly. In addition, the high-speed organization and the parallel processing which are embedded in the original self-organization also are realized in the new network. In this paper, some comparisons in which conventional neural networks, i.e., multilayer perceptron and self-organization models, are used also are shown. In particular, as a new application of speech ARM A modelling, a new spectrum envelope estimation is explained by using this network. [ABSTRACT FROM AUTHOR]

In the pulse transmission system such as pulse- position modulation (PPM), where the linear filter for pulse-timing detection is optimized and the construction of the detector is considered, one of the essential problems is the timing error contained in the timing information extracted from the received pulse. Several papers have been reported on the linear filters which are claimed to be useful in correcting the error, but the extent of improvement has not been analyzed clearly. This paper compares those filters from the viewpoint of the mean-square timing error and indicates the effectiveness of the filters on the relative basis. In addition, a new suboptimal filter is proposed and compared. Methods for filter optimization and timing detector construction are presented and compared from various viewpoints, such as the easiness of realization. The calculation of the timing error produced by the filter is calculated easily using the result shown in this paper, where the expression is normalized by the pulse-width. It is shown that the proposed suboptimal filter gives an excellent performance compared to the filters proposed in the past. Assuming the pulse transmission using the optical fiber, the calculation is shown based on the white and f2 additive noise as well as the Gaussian signal pulse. [ABSTRACT FROM AUTHOR]

This paper discussed the mathematical formation of the stochastic parabolic system, which is described by the multivalued operator called subdifferential operator, that plays an important role in the modeling of the free boundary problem such as viscoelastic and obstacle problems. For various deterministic distributed systems containing a subdifferential operator, the existence of the strong or weak solution is already shown. With respect to the stochastic distributed system, on the other hand, the existence of a strong solution has not been shown except for the system containing a subdifferential operator of a special convex function. As to the existence of the weak solution, there is a report of the case where the subdifferential operator appears in the interior of the system, but there is no report on the general stochastic system containing a subdifferential operator in the boundary condition. Another point is that a strong constraint is imposed on the initial value and the input function in the proof for the existence of the strong solution. From those viewpoints, this paper aims at ameliorating such a constraint and formulates the system by the weak solution. First, the existence condition for the weak solution is shown. First, the existence condition for the weak solution is shown. Then it is shown that there exists the maximal solution in the set of weak solution and an estimate for the maximal solution is derived which has a stronger significance than in the past formulation. [ABSTRACT FROM AUTHOR]

This paper discusses the hierarchical properties of digraphs and proposes flowchart decomposition algorithms using these properties. The hierarchical structure discussed here is based on only the topological relation- skips between vertices and the inclusive relationship among sets of vertices. When the given flowchart represents the control flow of a computer program, this hierarchical structure also shows strong correspondence between these inclusive relationships and program modules. The proposed flowchart decomposition algorithm consists of one algorithm for detecting modules and another algorithm for further decomposing the detected modules. This paper focuses on the former algorithm. Because the hierarchical structure discussed in this paper corresponds to the traditional program modules (e.g., function or procedure) of most of the computer programming languages, the module-detection algorithm can be used not only for decomposing flowchart but for more general purposes, including module design. Two types of module-detection algorithm are discussed in this paper. One detects all modules, and the other detects a restricted subset of modules from the standpoint of flow-chart decomposition. We then demonstrate with a practical example that the latter algorithm is ten times faster than the former. [ABSTRACT FROM AUTHOR]

This paper discusses the application of the error control techniques for parallel combinatory spread spectrum (PC-SS) communication system proposed previously by the authors. This paper considers the case where k-bit information is transmitted in the spreading sequences and considers the following two coding methods as suited to PC-SS systems: 1) to use 2k-ary Reed-Solomon code; and 2) to use the constant-weight code by Chung et al. in the transmitter, converting the data into the combinatory spreading sequences, and to suppress the error in the receiver by taking the double correlation. The bit-error rate performance is examined for the two methods and the effectiveness is discussed. It is shown as a result that both methods are useful in improving the bit-error rate of the PC-SS system. In the latter method, the number of spreading sequences needed per user is reduced to approximately one-fourth compared to the conventional PC-SS system with nearly the same bit-error rate. [ABSTRACT FROM AUTHOR]

This paper considers the problem of an optimum linear filter on the receiving side of the PPM transmission system in which both the additive noise and the signal-dependent noise exist. The optimum filter for the PPM transmission derived in this paper is the one in which the mean-square value (MSE) of the timing error is minimized at the location of the pulse detection. In contrast to the optimum filter in the PAM transmission which maximizes the signal-to-noise ratio in the amplitude direction, the present optimum filter is of a different type which maximizes this ratio in the time-axis direction. In this paper, the general equation which the optimum filter must satisfy is derived as an equation containing the previous research results. Subsequently, the timing error, optimum pulse detection level and the configurations of the filter on the receiving side and of the pulse location detector are presented. It is pointed out that, by the use of the optimum filter derived in this paper, baseband multiplexed transmission systems using pulses can be realized easily by combining the PAM and PPM systems. The method for multiplexing and the receiver configuration are shown. [ABSTRACT FROM AUTHOR]

This paper describes a design approach for constructing active R filters to simulate the doubly resistively terminated LC circuit and its compensation techniques. Using this approach, the necessary compensation resistance values can be obtained from only one measurement of the filter response. The approach in this paper is applied to an all-pole low-pass filter. However, it is shown that it can also be applied to a low-pass filter having transmission zeroes. [ABSTRACT FROM AUTHOR]

In this paper we consider a family of output waves obtained by impressing simultaneously an n-dimensional input wave f(X) to the finite number of time-Invariant linear net- works. A problem is considered, where the original wave f(X) is to be approximated using the sampled values of the forementioned output waves. A unified theory is presented with respect to the convergence of the approximate wave to the original wave f(X). The discussion given in this paper may be considered as a basis for the future theory of subband coding and multiplexing of the multidimensional signals, in the sense that we use the sampled values of the multidimensional waves transferred through the linear networks. The approximate wave h(X) for f(X) is a sum of the forementioned sample values of these output waves, being multiplied by certain n-dimensional waves called interpolation functions. The set of sampling points considered in this paper includes most of the practically import- ant arrangements of sapling points, such as hexagonal and octagonal lattices on the two- dimensional plane. The set of f(X) in this paper is defined as the set of n-dimensional waves such that the corresponding Fourier spectrum has the weighted powereofap norm (p > 1) which is less than a given positive value. Under several assumptions, sufficient, necessary, and necessary and sufficient conditions are derived for the wave h(X) always to converge to f(X). These convergence conditions are presented which are imposed on the transfer functions of the time-invariant linear networks through which f(X) propagates as well as the arrangement of the sampling points and the interpolation functions. They are closely related to the rank of the system of linear equations determined by those parameters. [ABSTRACT FROM AUTHOR]

The signal space composed of spline functions, which are smooth piecewise polynomials and almost free from oscillation, has attracted attention in the problems of interpolation and signal approximation. To make full use of such a signal space, there must be available a clear description of its properties. As a general approach to this problem, it is effective to derive the sampling basis in the signal space. Previously, the authors discussed the signal space composed of spline functions, and derived the sampling basis for the case where the degree is restricted to two. This paper extends the result to the arbitrary degree. Examples of sampling bases are shown for the practical cases of the second and the third degrees. The sampling basis derived in this paper provides the basic characterization of the signal space composed of spline functions of arbitrary degree. [ABSTRACT FROM AUTHOR]

In this paper we propose a method for the reciprocal use of knowledge between association rule extraction (a technique from data mining) and HMM-based information extraction. Association rule extraction is a method that is generally applied to structured data and is either inefficient or intractable when applied to data that lacks an accurate information schema. In this paper we use HMM-based information extraction to automatically recognize the attributes of each word in a set of textual data and assign tags in order to structure this data. In addition, we use the degree of association between words seen in the association rules extracted from the structured data to weight the HMM parameters in order to improve the recognition accuracy of the information extraction procedure and allow knowledge to be reciprocally shared between the data mining and information extraction procedures. Besides this we propose a data mining method that uses representative schema patterns found in different types of semistructured data such as tagged or BibTeX data using schema discovery methods. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 89(7): 51–60, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20256 [ABSTRACT FROM AUTHOR]

This paper considers an Internet agent system in which a tremendous number of agents operate, frequently appearing and disappearing, and discusses the fault-tolerant algorithm. Application of the snapshot algorithm to the agent system is considered. The snapshot algorithm is used to view the whole situation (snapshot) of the distributed system. The snapshot algorithm of Chandy and Lamport [2] is considered as a representative snapshot algorithm, in terms of the high efficiency and the simplicity of the procedure. It is not practical, however, to apply their snapshot algorithm to the distributed agent system in which a tremendous number of agents operate. From such a viewpoint, this paper extends the idea of Chandy and Lamport's algorithm and proposes a subsnapshot algorithm, in which the snapshot is taken among the agents who are in the causal relation, through message exchange and agent creation. Then, an efficient rollback algorithm is proposed, which is based on the snapshots taken by the subsnapshot algorithm. In the general rollback algorithm utilizing the snapshot, all agents must roll back. In contrast, in the rollback algorithm proposed in this paper, it suffices that only some agents should roll back. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(12): 43–57, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20208 [ABSTRACT FROM AUTHOR]

It is conceivable to embed additional information with a view to judge whether certificates existing as paper media or digital image data have been altered. The authors have already proposed and discussed a method for configuration of image information enabling reconstruction of the original image information from contaminated image information on paper by means of coding theory. Based on this proposal, a new method is being proposed for effective detection of alteration of digital image data by applying coding theory and the cryptographic theory at the same time. In the present paper, this method is critically discussed and a new method is proposed. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(8): 9–17, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20150 [ABSTRACT FROM AUTHOR]

In most of the methods of public key cryptography devised in recent years, a finite field of a large order is used as the field of definition. In contrast, there are many studies in which a higher-degree extension field of characteristic 2 is fast implemented for easier hardware realization. There are also many reports of the generation of the required higher-degree irreducible polynomial, and of the construction of a basis suited to fast implementation, such as an optimal normal basis (ONB). For generating higher-degree irreducible polynomials, there is a method in which a 2m-th degree self-reciprocal irreducible polynomial is generated from an m-th degree irreducible polynomial by a simple polynomial transformation (called the self-reciprocal transformation). This paper considers this transformation and shows that when the set of zeros of the m-th degree irreducible polynomial forms a normal basis, the set of zeros of the generated 2m-th order self-reciprocal irreducible polynomial also forms a normal base. Then it is clearly shown that there is a one-to-one correspondence between the transformed irreducible polynomial and the generated self-reciprocal irreducible polynomial. Consequently, the inverse transformation of the self-reciprocal transformation (self-reciprocal inverse transformation) can be applied to a self-reciprocal irreducible polynomial. It is shown that an m-th degree irreducible polynomial can always be generated from a 2m-th degree self-reciprocal irreducible polynomial by the self-reciprocal inverse transformation. We can use this fact for generating 1/2-degree irreducible polynomials. As an application of 1/2-degree irreducible polynomial generation, this paper proposes a method which generates a prime degree irreducible polynomial with a Type II ONB as its zeros. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(7): 23–32, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20151 [ABSTRACT FROM AUTHOR]

Due to requirements for large-capacity data communication, the bandwidth of wireless communication systems such as cellular phones and wireless LAN has been increased. In this paper, the recent trends of wireless communication systems are reviewed with an emphasis on the receiver analog front end. In addition, a design technique for wide-band circuits is described, with the mixer in the receiver as an example. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(4): 1–10, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20146 [ABSTRACT FROM AUTHOR]

In this paper, we examine what type of bifurcation develops when an extremely small periodic force drives a van der Pol oscillator that generates the duck solution. We create the bifurcation diagrams and explain the structures of bifurcation in the fundamental harmonic entrainment region, 1/2 subharmonic entrainment region, and 1/3 subharmonic entrainment region. In each region, we observed the cascade generation of the period-doubling bifurcation in the extremely small periodic force range, and the occurrence of chaos. We also discovered the very interesting phenomenon of the existence of two duck solutions when driven by an extremely small periodic force. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(4): 51–59, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20093 [ABSTRACT FROM AUTHOR]

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]

This paper proposes a relaxation algorithm for an inverse filter that is adaptive to environmental change in a multichannel sound reproduction system. For a conventional multichannel sound reproduction system, a design method has been proposed for an inverse filter by means of the Moore–Penrose-type generalized inverse matrix. However, in the inverse filter in the conventional method, background noise and line noise may be enhanced when the room transfer characteristics vary. In order to resolve this problem, an adaptive relaxation method based on the truncated singular value decomposition (TSVD) has been introduced. In the present paper, a microphone for monitoring the noise is placed at a location far from both ears of the listener. Relaxation of the inverse filter is carried out by using the reproduced sound observed by this microphone. From the results of computer simulation, it is found that the proposed method prevents amplification of the noise. Also, as a result of a subjective evaluation experiment on sound reproduction in a real acoustic environment, the sound quality was found to be improved significantly without damaging the sound localization of position. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(7): 15–26, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10159 [ABSTRACT FROM AUTHOR]

This paper considers chaotic masking and reports on the amplification of the information signal, which is observed in restoring the information signal, together with its mechanism. The systems considered in this paper are the R-L-Diode circuit, the Lorenz system, and the generalized Hopfield network. It is found that the result of restoration depends on the frequency of the information signal, and it is intended to account for the amplification phenomenon by examining the frequency characteristics of these systems. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(1): 89–95, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1134 [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]

In this paper, we study improving the noise robustness of formant frequency extraction based on linear predictive analysis by using the noise reduction ability of the autocorrelation function. The autocorrelation function has the property of concentrating the noise components in the short delayed parts of speech signals corrupted by white noise. We believe that the noise robustness of formant frequency extraction can be improved by utilizing the above property and using the autocorrelation function of the speech signal instead of the signal itself in formant frequency extraction. In this paper, we analyze the principle behind extracting the formant frequency from the autocorrelation function and examine this problem and a possible solution in detail. Then based on this analysis, we propose Method 1 that performs linear predictive analysis of the autocorrelation function of the speech signal. Then we verify from the experimental results that an extraction accuracy at the same level as the conventional method is obtained for a clean signal by Method 1, and the extraction accuracy is significantly improved over the conventional method for speech corrupted by noise. However, inadequate extraction accuracy is also indicated in a very noisy environment, and we analyze the cause and propose Method 2 as an improved method that subtracts the autocorrelation function. The experimental results show that when the signal-to-noise ratio is 15 dB or less, the formant frequency extraction error (Average Absolute Error) in Method 2 is kept to about one-third the error of the conventional method and about one-half the error of Method 1. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(9): 1–14, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1114 [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]

FIR wavelet filters are mainly used in conventional wavelet-based image coding. However, FIR filters, except for the Haar wavelet, cannot satisfy both the exactly linear phase and orthonormality conditions. For example, the well-known Daubechies-9/7 wavelet is biorthogonal. In this paper, an effective implementation of the all-pass-based wavelet filters is presented for wavelet-based image coding. Since all-pass filters are IIR, both the exactly linear phase and orthonormality conditions can be simultaneously satisfied, so that better compression performance can be expected. Also, IIR filters can be realized with a lower computational complexity than can FIR filters. Finally, the proposed IIR wavelet filters are evaluated by using SPIHT to compress the practical images, and the influence of the all-pass filters and delay elements on the compression performance is investigated. It is shown through the experimental results that the IIR wavelet filters proposed in this paper have a lower computational complexity than the Daubechies-9/7 wavelet, with a comparable compression performance. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 85(2): 13–21, 2002 [ABSTRACT FROM AUTHOR]

Checkerboard distortion is the kind generated in the sampling rate converter using a digital filter. In this paper, such distortion is studied in the continuous-time domain. The distortion is newly defined in the problem of interpolation by a continuous interpolation function and conditions for avoiding the checkerboard distortion are derived. Further, it is shown that a digital filter that does not generate checkerboard distortion can be created by an appropriate sampling to the continuous interpolation function satisfying the conditions in this paper. The relationship between the continuous interpolation function with time scale transform and the checkerboard distortion is studied. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(12): 59–68, 2001 [ABSTRACT FROM AUTHOR]

An upper bound or a lower bound of the Multiple-Choice Knapsack Problem can be calculated by solving LP relaxation. In 1979, Sinha and Zoltners proposed a branch-and-bound algorithm for solving the Multiple-Choice Knapsack Problem, and provided a method to obtain the strict upper bound. In this paper, we propose a new calculation method to obtain a stricter upper bound than the upper bound of Sinha–Zoltners and compare the results of the two methods. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(7): 22–27, 2001 [ABSTRACT FROM AUTHOR]

We propose a new model of texture images using morphological filters. This method is based on a linear predictive model [autoregressive (AR) model], which is a representative texture image model. This method embeds the frequency characteristics of the texture image in the AR model. In this paper, we will demonstrate a new model that represents the spatial features of a texture image. Morphological filters can remove the spatial features and are used for pattern spectra. In this paper, we consider embedding the spatial features in the structuring elements of morphological filters and propose an Opening Model (OM) to model the convex up features and a Closing Model (CM) to model the convex down features. We examine the reliability of the proposed model and present improvements to the low-reliability model. Furthermore, our recognition tests clearly show excellent recognition when OM and CM are used together. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(7): 53–62, 2001 [ABSTRACT FROM AUTHOR]

The minimax approximation is of special importance for the design of filters. As to the minimax approximation by a rational function, the characteristics of the best approximations are well developed. However, least-squares approximation is often used instead due to the simplicity of calculation. Nevertheless, no theoretical result has been published as to how close and how good the achieved approximation results are compared to those of the best (optimal) minimax approximation. This paper deals with the least p-th approximation (p ⩾2, even) by a rational function and gives a fairly simple theoretical lower bound for the ratio of the maximum error(s) of the minimax approximation to those of the least p-th approximation (usually local minima). This paper proves that on a weak but practical assumption, Nishi's result on the least p-th approximation by linear function is also valid for the approximation by rational function. Numerical examples show that the least p-th approximation for p = 8 or 16 is usually enough to achieve a good approximation to the minimax approximation. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 84(3): 21–32, 2001 [ABSTRACT FROM AUTHOR]

This paper considers the Alpazur oscillator, which is composed of a Rayleigh oscillator unit and a dc power supply controlled by a switch, and examines the bifurcation set by numerical calculation. In such a case, a bifurcation diagram is constructed in order to see the phase-space behavior of the system. In the conventional method, however, the calculation is possible only when the trajectory is continuous and differentiable. The method cannot directly be applied to the case where the trajectory is discontinuous or not smooth at a point (not differentiable) as in the case of on–off circuits. From such a viewpoint, this paper proposes a bifurcation trajectory tracking algorithm for the that nonlinear on–off circuit, containing a switch that is made on and off periodically by an external force. As the method of analysis, the composite Poincaré mapping is defined based on the periodic operation of the switch. The method of calculating the bifurcation parameter is described precisely. Using the proposed method, the Alpazur oscillator is analyzed. The phase-synchronized solutions that exist along the Neimark–Sacker bifurcation as well as the chaos are investigated. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 84(1): 75–83, 2001 [ABSTRACT FROM AUTHOR]

In this paper, we propose an online design method of a nonlinear inverse system which eliminates the linear and nonlinear distortions of the nonlinear system at the same time. We discuss the online design method by using the adaptive Volterra filter. Although we have already proposed offline design methods, these had the problem that the elimination level of the nonlinear distortion degrades when the characteristics of the target nonlinear system vary. In this paper, we first show the effects of compensation of distortions due to the modeling error of each filter which composes a nonlinear inverse system. Next, the proposed online design method is explained by dividing it into three steps. In the first step, the online identification method of the first-order Volterra kernel is discussed. In the second step, the online design method of the linear inverse filter is discussed. In the third step, the online identification method of the second-order Volterra kernel is discussed. Finally, the effectiveness of the proposed method is shown by computer simulations. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(9): 46–56, 2000 [ABSTRACT FROM AUTHOR]

This paper considers waveform observation by the normal-type natural observation transform, and discusses the instantaneous relation between the waveform to be observed and the normal-type basic observed value sequence. The relation is similar to the uncertainty relation in the Fourier transform. Although the global uncertainty relation in the normal-type natural observation transform is clearly formulated already, the result is still insufficient for the theory of natural observation, where the instantaneous handling of the waveform has to be formulated. From such a viewpoint, the relation shown in this paper is significant, as a basic property in the instantaneous handling of the waveform by the theory of natural observation. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(8): 61–70, 2000 [ABSTRACT FROM AUTHOR]

A lossless coding method that can restore digital images without any distortion is needed in areas that rely on high-quality images such as medical and broadcast applications. Lossless subband coding can obtain a high compression performance and enables progressive image transmission, and hence is known to be an effective lossless coding method. In the conventional two-band lossless subband coder, there are restrictions in the filter tap length, filter coefficients, and type of filter bank such as separable or nonseparable. Therefore, it is difficult to design the subband coder flexibly in response to special needs such as to increase the compression efficiency for a specific image, or to reduce the distortion in the progressive image transmission. In this paper, a two-band lossless subband coder is proposed that has several degrees of freedom in selection of the filter tap length, filter coefficients, and type of filter bank. These degrees of freedom are used for improvement of compression efficiency for a specific image. The filter coefficients are determined by the nonlinear programming. The proposed method is a general form containing the conventional method. For several test images, a compression efficiency higher than that in the conventional two-band lossless subband coder is confirmed. © 1999 Scripta Technica. Electron Comm Jpn Pt 3, 83(4): 103–114, 2000. [ABSTRACT FROM AUTHOR]

This paper proposes high-level synthesis using a genetic algorithm that can realize register transfer level circuits from behavioral descriptions. Until now this type of synthesis problem has been separated into scheduling and allocation problems, and the two problems have been solved separately. This procedure has a difficulty that the process may easily fall into a local solution.This paper proposes a simultaneous solution of those problems, so that a circuit better than in the conventional algorithm can be synthesized. The proposed approach, on the other hand, has the disadvantage that the computation time tends to be longer, since the length of the chromosome is increased. To avoid this difficulty, several measures are used such as modification of a faulty gene and discrimination among circuits requiring the same number of resources. A result better than in the conventional approach is obtained for the benchmark problem. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(4): 24–32, 2000 [ABSTRACT FROM AUTHOR]

Image resolution conversions are methods for converting a digital image to another with a different number of pixels. In this paper, two image resolution conversion methods in the DCT region for image minification are discussed. The images minified by these methods are known to have different picture quality. Specifically, one of them has the advantage of excellence in realization while the picture quality is known to be poor. However, the cause of the differences in picture quality has not been found. In this paper, the equivalent representations of these conversion methods in the time domain are derived. Based on these equivalent representations, the cause of the difference in picture quality in the two methods is theoretically explained. Further, by extending the conventional method, a generalized image resolution conversion method in the DCT domain is proposed. As a result, it becomes possible to improve the picture quality in the conventional method. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(3): 31–41, 2000 [ABSTRACT FROM AUTHOR]

This paper studies the linear cryptanalysis of the MBAL (MultiBlock ALgorithm) Cipher. MBAL includes SXAL8 Cipher and 3-round Fm function. Moreover, the size of the input data is variable. In this paper, we propose the strategy of making linear approximations for MBAL which are independent of SXAL8 and the size of the input data. We succeed in making an approximation with bias 2–5.49 from the first to second (or second to third) rounds, where the bias is defined as |Prob – 1/2|. Next, we show how to reduce the number of “effective key bits” from 96 to 80 in the first (or n-th) round. Finally, we succeed in making an approximation for MBAL with bias 2–16.57, which gives 1 bit of information on the key. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(10): 1–8, 1999 [ABSTRACT FROM AUTHOR]

This paper derives a renormalization formula defined on the parameter space where mapping behavior is preserved, together with the equivalent potential function. In contrast to the universal function given by Feigenbaum, the behavior near the critical point is governed by the potential function. There are several interesting features which are shared by Feigenbaum's universal function, such as the representation of the critical point in terms of the unstable fixed point of the potential function, but the mapping differs from the scaling. The one-dimensional mapping is considered as an example, and the critical point and the scaling, which are major constants characterizing chaos from the potential function, are calculated precisely. © 1998 Scripta Technica, Electron Comm Jpn Pt 3, 81(8): 41–51, 1998 [ABSTRACT FROM AUTHOR]

When a planar graph is drawn, the drawing plane is divided into several regions, each of which is called a face. A subgraph composing the periphery of a face is called a window. For each window W that exists in the drawing of a planar graph G, the total sum of the weights of the vertices and the edges is called the weight of the window W. Among all windows that can appear in drawings, the window with the maximum weight is called the maximum-weight-window of C. This paper considers the problem of determining the drawing with the maximum-weight-window. The result for this problem is already obtained for the case where the graph is 2-connected, as well as for the case where the graph is not necessarily 2-connected but all weights are nonnegative. This paper considers also the arbitrary planar graph with real-valued weights, and shows that the drawing with the maximum-weight-window can be determined in O(n + e) time. This result can be applied to the layout design for LSI and printed board, as well as to the diagram representation of the network. It can also be applied to the solution of the maximal planarization problem for the graph. [ABSTRACT FROM AUTHOR]

The adaptive line embracer eliminates noise from a sinusoidal signal with the unknown frequency buried in the noise and extracts the sinusoidal signal. The performance of the adaptive line embracer is represented by the signal-to-noise improvement figure. The signal-to-noise improvement figure increases as the bandwidth of the adaptive line figure increases is made narrower. The convention adaptive line enhancer, however, contains an incompatible problem in that the convergence of the adaptive coefficient is deteriorated when the signal-to-noise improvement figure is by narrowing the bandwidth. This paper proposes the adaptive line enhancer with the discriminated structure to solve this problem. The adaptive line enhancer with the discriminated structure is composed of two parts. One is the (frequency) estimator that estimates the frequency of the unknown sinusoidal signal, and the other is the (sinusoidal signal) discriminator that discriminates the unknown sinusoidal signal from the noise. The estimator is the conventional adaptive line enhancer, and the bandwidth is determined so that the convergence of the adaptive coefficient is the fastest. The discriminator is the narrowband bandpass filter where the center frequency is controlled by the adaptation coefficient. This paper the optimal design method for each of those paths, and the effectiveness of the method us verified by a computer simulation. [ABSTRACT FROM AUTHOR]

The word communication in the computer science field is associated with data communication. In the original context, communication is "person-to-persons communication" using some language or the human voice. Nowadays, handicapped people are assisted in communication by informational equipment. The authors have developed several tools as welfare communication tools for handicapped people and related volunteers. The main theme is a support system for the visually impaired and blind. Not only is research level equipment developed, but also commercial products. In this paper, several are described, along with some problems encountered in the development of welfare communication and accessibility to handicapped people. [ABSTRACT FROM AUTHOR]

Recently, quality improvement of housing and the living environment has become a priority issue. The quality improvement of the acoustic field in a direct voice transmission as a fundamental method of information transmission is no exception. It is important to realize an acoustical environment in which the audience can concentrate on hearing sound without distraction due to external noise. The relationship between the voice signal and the external noise has been studied extensively by a number of researchers, thus significant achievements have been reported. Most of the results, however have more or less focused on the voice signals from the viewpoint of perception and understanding of the voice with respect to the masking effect and articulation. This paper differs from these studies in the following two aspects. First, the intent is to evaluate the external noise instead of the voice signal. Second, the sensory evaluation is based on the subjective judgment of human beings. The fuzziness necessarily associated with the subjective quantity is taken into account from the outset. Specifically, by using the fuzzy theory, this paper studies the problem of estimating or predicting the psychological judgment of the audience when meaningless nonwhite noise with an arbitrary power spectrum eaten externally.The validity of the method proposed here is verified by means of the measured data from an indoor psychological experiment. [ABSTRACT FROM AUTHOR]

This paper shows that two complex coefficient transfer functions can be realized by a single hypercomplex coefficient network, utilizing all four outputs of the hypercomplex network. The realization is considered for the case where the two arbitrary complex coefficient transfer functions do and do not have a common denominator. It is shown that the transfer functions can he realized by the hypercomplex coefficient network of the same order in the former case and by the hypercomplex coefficient network with the half-order in the latter case. The realization of the transfer functions with the common denominator is interesting, but the general method to derive the transfer functions is not known. From such a viewpoint, this paper proposes the discrete-type Kautz approximation with the positive frequency as the domain of approximation, including the optimal pole determination algorithm. By this approach, two complex coefficient transfer functions with an n-th-order common denominator are derived from the two 2n-th-order real or complex coefficient transfer functions. The method can be considered as one that simultaneously approximates the amplitude and the phase. The proposed approximation method is an approximation only for the positive frequency domain, and the property of the analytic signal that the negative frequency component is zero can be utilized. As a result, the method proposed in this paper has the advantages that the degree of freedom in the filter design is enhanced and the sampling frequency can be halved. In addition, the network structure can be simplified. [ABSTRACT FROM AUTHOR]

This paper considers the oscillators coupled in the form of a ring through resistors. It is described for the case of five oscillators that, for the symmetrical solutions, the analysis based on the conjugate class in group theory is useful. Major problems in tie coupled oscillators are the existence of the synchronized periodic solution and its stability. In the case of the weakly nonlinear system, the problems of the synchronized periodic solution and the stability are almost answered by techniques that include mode analysis. In the case of the strongly nonlinear system, on the other hand, the periodic solutions with symmetry have been classified mostly based on the subgroup in the group theory when the number of coupled oscillators is less. When the number of oscillators is increased, however, the number of subgroups becomes tremendous, which makes systematic analysis difficult. This paper uses the conjugate class and shows that tie symmetrical solutions can be classified systematically. It is also shown that the approximate periodic solution with symmetry can be approximately determined by the averaging based on the symmetry. [ABSTRACT FROM AUTHOR]

The solution of a system of linear equations is very important in various fields of engineering, and various methods have been presented for the numerical solution. Especially, in the adaptive signal processing and adaptive control, the system of equations is often considered where the coefficient matrix is (pseudo-) positive-definite and symmetrical. The conjugate graduate method is known as a means to solve such a system of equations. Letting the rank of the coefficient matrix be R, it is a method where the solution for the system of equations is determined by iterating a certain procedure R times. In this method, even if the iterative process is terminated on the way due to the constraint for the computation time, for example, an approximate solution which is somewhat accurate is obtained. This method, however, still has room for improvement with respect to the accuracy of the approximate solution. From such a viewpoint, this paper examines geo- metrically the iterative computation process in the con- jugate gradient method and, based on that result, propos- es a new method of iterative solution for the system of linear equations with symmetrical coefficient matrix. Then the original method is slightly modified and an adaptive algorithm is derived by applying the result to the parameter estimation of the adaptive filter. Those methods are based on the orthogonal projection operation from the solution to be determined to the linear space. Consequently, it is expected that, even if the iterative computation is terminated on the way, a more accurate approximate solution is obtained, compared to the case of the conjugate gradient method or the adaptive algorithm based on the conjugate gradient method. By the property of the method proposed in this paper, the method is useful when the coefficient matrix is ill-conditioned. It is expected that the method will effectively be utilized in the signal processing and control. [ABSTRACT FROM AUTHOR]