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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]

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]

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]

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]

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]

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]

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 discusses the even-length real-valued cyclically orthogonal pseudonoise (PN) sequences with small absolute values as the PN sequence for spread spectrum communication. First, the sequence of maximum absolute value 2 with cubic or inverse phase parameters is considered. By replacing the sequence variable by the power of the primitive root and by regarding the power as the sequence variable, an orthogonal PN sequence is obtained. There exists the same number of such sequences as the number of primitive roots, and the crosscorrelation function is relatively small if the period is large. By convolving a sequence with itself, a set of sequences is obtained which has relatively small absolute value of the sequence and relatively small absolute value of cross correlation. It is proposed also to convolve a sequence with other sets of sequences. The sequence considered in this paper is suited to the digital correlation processing using the fast-Fourier transform when the period is 4, 16, 256, 65,536, ... . [ABSTRACT FROM AUTHOR]

The relaxation method is one of the methods for solving an initial value problem for a set of differantial equations and is suitable for analysis of large-scale circuits such as those simulating VLSI circuits. In analysis of the relaxation method, time domain analysis is most common but complex frequency domain analysis by means of the Laplace transform also is applicable. This paper presents the Laplace-transform analysis. Semiconducting elements such as transistors of the VLSI circuits exhibit nonlinear characteristics. Thus, the differential equations to be used in the relaxation method must be nonlinear. In some cases, the properties of nonlinear dements are given not by analytic functions but by a series of discrete numerical values such as observed data. To express the nonlinearity of these elements, multidimensional interpolation formulas are used in this paper and practical computer algorithms for solving them are derived. The analysis of the conditions of convergence of the relaxation method shows that the proposed algorithm is applicable to circuits which operate in both stable and unstable domain, such as oscillators. [ABSTRACT FROM AUTHOR]

This paper considers the transfer function containing several kinds of variable parameters and proposes a method for synthesizing the circuit by the state-space method. The method is to construct separately the denominator and the numerator of the given multiparameter transfer function. Using the proposed method, the multiparameter transfer function can be constructed by the active RC circuit with the minimal number of dynamical elements and about half the number of variable elements, compared with the traditional method. A method is also shown which utilizes free parameters of the circuit structure to minimize the pole-zero sensitivity of the multiparameter transfer function with reset to the variation of the active and passive elements. Finally, an illustrative example is shown to demonstrate the effectiveness of the synthesis method proposed in this paper. [ABSTRACT FROM AUTHOR]

This paper proposes a method for precisely estimating sinusoidal frequency using five samples. The estimation procedure is implemented by interpolating the impulse location between DFT results. Five samples are used because two complex components are available in the DFT results. However, an estimation method which uses four samples and is called Prony's approximation method is known in the literature. To derive interpolation formulas for six or more samples is more complex. In this paper, strict estimation formulas and their simplified versions arc deduced from the DFTs of five samples. The frequency estimation error is also discussed with a computer-simulated experiment which shows a I percent r.m.s. error at a signal-to-noise ratio of 30 dB. This method may be applicable for representing frequency in a direct binary form which can be processed by simple operations such as addition, exclusive-OR, and sign extractions form the calculated value. [ABSTRACT FROM AUTHOR]

This paper proposes a method of realizing an adaptive normalized ARMA lattice filter. This filter was proposed in a previous paper and can be realized with an arbitrary ARM A order and with all the lattice coefficients less than one. Furthermore, these coefficients are equal, so the structure is simplified. The adaptive realization method proposed in this paper retains these features. In addition, we perform spectral estimation of a time-varying signal with frequency weighting and model identification of an unknown input system. In our paper we describe an experiment with a real speech signal to prove that our algorithm can be used for frequency-weighted spectral estimation. The use of real data to perform spectral estimation has not been attempted in other references. [ABSTRACT FROM AUTHOR]

This paper presents an optimization procedure using tile stochastic model function, aiming at the application to the optimization of the circuit constant in the integrated circuit. The optimization of the circuit constant is formulated as a minimization problem for the objective function derived from the circuit response under the specified condition. In the method presented in this paper, the model function is constructed which approximates well the objective function, and the objective function is minimized by minimizing the model function. The features of the method are that the algorithm is simple, the number of evaluations of the objective function is small, the result depends less on the initial value, and the minimum of the objective function can be determined with a high accuracy. The method is evaluated using the standard test function, and it is verified that all minimum values of the objective function can be determined with n high accuracy with fewer number of calculations of the objective function. The optimization procedure is applied to two circuit examples: the analog control circuit and the inverter chain. The result is satisfactory and the usefulness of the presented optimization procedure in the circuit design is demonstrated. [ABSTRACT FROM AUTHOR]

This paper proposes a new method of synthesis for a CMOS LSI-oriented asynchronous sequential circuit with the two-transistor AND logic structure, using the one-hot code for the internal state assignment. The assignment of the state by the one-hot code is executed uniquely, independently of the properties of the state-transition table. The CPU time for state assignment, which has been a problem, can be ignored. The critical race-free property, which is then a problem, is solved by a 4-FET set/reset flip-flop which stores the stale variable and the drive circuit for the flip flop, not the internal code assignment. The set and the reset driving circuits have the AND logic circuit structure, where two transistors are placed (with the state variable and the external input as the gate inputs) on each conduction path. The synthesized circuit is hazard-free. It realizes a pseudo-CMOS operation without static power consumption. Comparing the method proposed in this paper with other methods of synthesis, it is compact and especially suited to the large-size asynchronous sequential machine (with 40 or more cells in the row-column product of the state transition cable). [ABSTRACT FROM AUTHOR]

This paper considers the software error detection process in the testing phase, which is the final stage of the software development. The process is modeled by a stochastic process with a continuous changing variable. Employing a new mathematical technique based on the stochastic differential equation, a new software reliability growth model is constructed. First, the traditional basic differential equation which describes the average behavior in time of the number of errors, is extended to the Itô-type stochastic differential equation, and the probability distribution of the solution process is investigated. Then, using the derived distribution, several reliability assessment measures are derived. Finally, the unknown parameters of the model are estimated based on the actual test data, and the stochastic property of the temporal variation of the errors remaining in the software is examined numerically. Using MTBF (mean-time between failure), which is the important reliability assessment measure, the problem to determine the release time of the software is discussed. [ABSTRACT FROM AUTHOR]