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Recently, occurrences of chaos have been reported in many engineering systems and most of these occurrences are unfavorable phenomena which used to be controlled. This paper proposes two methods for the stabilization of arbitrary unstable periodic orbits embedded in chaos attractors existing in nonlinear discrete-time systems using nonlinear feedback. First, we propose a method based on a property of contraction mappings. Next, we propose another method using exact linearization via nonlinear feedback. In both methods, differentiability of the controlled systems is not required. Moreover, feedback need not be reconstructed in target periodic orbits. [ABSTRACT FROM AUTHOR]

This paper considers the complex coefficient transfer function with transmission zeros on the imaginary axis and presents a method to realize such a transfer function by a reactance network including imaginary resistors with double resistor termination. Using the obtained network and the reference network for the complex-wave digital filter, the complex digital filter with transmission zeros on the imaginary axis can easily be extended, which is important in practice. In this paper, Brune's method modified by Belevitch as well as Fujisawa's method extended by Morimoto et. al. are improved and the realizability condition is relaxed. Then, several construction examples are shown an the features of construction methods are compared and discussed. [ABSTRACT FROM AUTHOR]

This paper studies the chaotic oscillations in a dissipative system generated by a second-order direct-form II digital filter with a 2's complement adder. The locus of this chaotic oscillation is absorbed into the parallel lines comprising the Cantor-like set on the state space. In this paper, with respect to the chaotic oscillation, the initial value dependence of the locus, the output distribution, and the fractal dimensions of the locus on the state space are investigated as the fundamental nature of the chaos. Also, by considering this filter as the two-dimensional mapping, the mechanism generating the Cantor-like attractor is studied. From this study, it is shown that the 2's complement adder characteristic has functions for stretching and folding of the state space causing the chaos. Also, the coefficient dependence of the Cantor-like attractor is studied and the relationship between the fractal dimensions of the locus and the coefficient are presented. [ABSTRACT FROM AUTHOR]

When no function symbols occur in a first-order predicate formula, there exists a procedure to decide the satisfiability of the formula. Examples of such procedures are the method based on Robinson's resolution principle and the method based on the partial instantiation technique by Jeroslow. This paper considers the case where the formula is clausal form and no function symbols occur in it and proposes a satisfiability decision procedure based on the partial instantiation technique applicable to the considered case. The procedure is based on the partial instantiation technique, as in Jeroslow's method, but aims at the following aspects by utilizing the formula being in clausal form: (1) Davis-Putnam's method is used in solving the satisfiability problem in the prepositional logic; (2) the rapid expansion of the formula can be suppressed by expanding the formula with the clause as the unit; and (3) the number of unifications is reduced. The proposed method is compared to Jeroslow's method and the method based on the resolution principle, and then the effectiveness is demonstrated. [ABSTRACT FROM AUTHOR]

The overlap-add (OLA) and overlap-save (OLS) methods are well known as efficient schemes for higher-order filtering. In those methods, the linear convolution for the infinite-duration data is reduced to the circular convolution for the finite blocks, and the fast-Fourier transform (FFT) is applied to the execution of the circular convolution. This paper proposes new OLA and OLS methods, which can directly be applied without a redundancy to the rate conversion. First, the conventional theory is extended for the finite-duration data. Then, the result is generalized for the infinite-duration data. Two methods are proposed as the main result in this paper, which are the extended overlap-add and extended overlap-save methods. Finally, the proposed method is compared to the polyphase structure, which is well known as an effective realization method for the rate conversion. [ABSTRACT FROM AUTHOR]

One of the important features that characterize the physical random field or the image texture is Taylor's correlation length. However, there has not beet presented a clear-cut theory as to the foundation of the correlation length from the viewpoint of probability theory or statistical theory. Another aspect is that the feature is one-dimensional, and, therefore, the application to the random field is limited to the isotropic random field. This paper defines the nth order statistical number of degrees of freedom as the equivalent number of uncorrelated points in the square finite region in the homogeneous random field. From the definition, the correlation area and the correlation length are derived. The correlation area is referred to as the average area of the region statistically affected by an arbitrary point on a two-dimensional field. The correlation length represents the effective distance between adjacent independent or uncorrelated points on the line parallel to the x- or y- axis. The first-order correlation length is the same as the conventional Taylor correlation length. The first-order correlation area is a natural extension of Taylor's correlation length to the two-dimensional random field. The correlation length along the principal axis, which is independent of the definition of the coordinate axis, is also derived. Those features are based on the theory of statistics and are expected to be useful for the analysis of the actual random physical field and the texture analysis of the image. [ABSTRACT FROM AUTHOR]

This paper notes the fact that the recording code often takes the form of a multitrack code, as well as the fact that the k-constraint in the (d, k)-constraint is considerably relaxed, which increases the capacity. Then the paper proposes a systematic construction algorithm for the fixed block-length multitrack (d, k) code and discusses the coding characteristics. A simple idea is applied in the code construction to apply the (d, ∞) code to all tracks except for one in order to reflect the relaxation of the k-constraint on the improvement of the coding rate. The (d, ∞) code is applied to the left track if the vector obtained by the elementwise logical sum of the code words satisfies the k-constraint and, if not, the (d, k) code is applied. The author has already proposed a systematic construction algorithm for the efficient fixed block-length (d, k) code. This paper discusses mainly the systematic construction algorithm and the coding characteristics for the fixed block-length (d, ∞) code, based on the unique ordering technique of the binary sequence by the Schalkwijk algorithm, as the systematic construction algorithm for the fixed block-length multitrack (d, k) code based on the forementioned idea. The coding rate and the coding efficiency are examined, and the result indicates that the characteristics are very close to those of the (d, ∞) code. [ABSTRACT FROM AUTHOR]

The information storage algorithm of a Hopfield-type network is a kind of self-correlation learning. This algorithm needs the condition in which vectors of learning patterns cross at right angles to each other. This paper proposes n method to make this condition easier. To this purpose, threshold is used. An investigation of how to decide quantity is included. This paper discusses a network with units whose state is 1 or 0. Fixing method of threshold has not been proposed before to this type of network. It is, therefore, proposed in this research; the goal is to facilitate learning patterns. [ABSTRACT FROM AUTHOR]

This paper proposes a new speech coding method pitch synchronous innovation code excited linear predictor (PSI-CELP). This method is based on CELP but adds pitch synchronous innovation. This results in even random code vectors being adaptively converted to have pitch periodicity for voiced frames. This scheme can improve the synthesized speech quality of voiced frames in the low bit-rate CELP without increasing either computational complexity or bit rate. In addition to pitch synchronous innovation, this paper also proposes four other methods in which the quality of synthesized speech is effectively improved or the computational complexity is reduced for low bit-rate CELP: (1) the perceptual weighting filter is an MA-type and uses nonquantized LPC parameters; (2) the fixed codebook is prepared as part of the adaptive codebook for nonperiodic frames; (3) the random codebook consists of two-channel conjugate sub-codebooks; and (4) codebook searches are made using delayed decision. In a speech coding system with pitch synchronization of random code vectors, improvements in ratio SNR, segmental SNR and opinion equivalent Q value are respectively 1.6 dB. 1.0 dB. and 0.5 dB. These basic methods were adopted in the standard code for Japanese half-rate digital mobile telephone service. [ABSTRACT FROM AUTHOR]

This paper proposes a new method for vector quantization (VQ) of LSP parameters, using moving average (MA) interframe predictions. MA predictive VQ executes the prediction using the code vectors (codebook outputs) in the previous frames. Moreover, this method has the following features, compared to autoregressive AR) predictive coding, which executes the prediction based on the previous quantized (decoded) values: (1) even if a bit error is produced in the transmission channel, its effect to the succeeding frames remains finite; and (2) the stored codes can be decoded from any time point. This paper discusses the quantization performance and the robustness against bit errors of the MA predictive VQ, as well as the training method for the codebook. Assuming an application to low-bit-rate speech coding, the configuration considered has a frame length of 40 ms, with each frame composed of four subframes. A method for further improvement of the efficiency is reported. As a result of evaluation using actual speech data, bit reductions of approximately 16 percent and 23 percent are achieved for the quantization for each 20 ms and 40 ms, respectively, compared to the conventional method which does not use interframe prediction. [ABSTRACT FROM AUTHOR]

With regard to the theory of natural observation proposed as a new method of waveform analysis and although papers by the author have already been published over the last few years, this paper attempts a generalization based on the theory. As a result, the theory of natural observation can largely be divided into the neighboring type and the equilibrium type. Further, it was clarified that the previous theories belong to the neighboring type. In the treatment of acoustic vibration waveforms such as voice, the hitherto not discussed equilibrium type is desirable. Also, based on the forementioned theories, the general form of natural observation transformation is proposed anew as a substitute for Fourier transformation and, in particular, the standard form thought to be useful hereafter is shown concretely. [ABSTRACT FROM AUTHOR]

The theory of natural observation has been proposed as a new method of analysis to replace the Fourier analysis method. It was systematized as a theory with two major types-the neighboring type and the equilibrium type. Especially, although the nucleus of the theory is of standard form, the description with regard to the starting point of the theory in the past has not always been plain and comprehensible. This paper is a radical improvement in this regard and was written to clarify the origin of the theoretical development. Although the previously published theories are not completely revised, this paper is considered significant from the viewpoint of clarification of the base of the theoretical development. [ABSTRACT FROM AUTHOR]

Often, in handwriting a character, a natural expansion or contraction occurs even if the writer intends to write the character in a specified size. This paper considers the expansion-contraction phenomenon which appears statistically in drawing a figure in general. A theoretical investigation is presented from the theory of probability, and the shape of the probability distribution governing the phenomenon is analyzed mathematically. The approach also gives an answer from the viewpoint of the theory of probability to the problem where the length is expanded or contracted unintentionally, even if a constant length is intended. This paper shows a typical mathematical approach in which the determination of the shape of the probability density function is reduced to the solution of the special functional equation. It is considered to be interesting also from the viewpoint of the theory. [ABSTRACT FROM AUTHOR]

Recently, many applications of integer theory to cryptographic techniques have been observed. The discrete logarithm problem is one such case. Usually, the discrete logarithm problem is the determination of the logarithm for the given arbitrary element with a prime number as the modulus. However, the discrete logarithm problem can also be considered with a composite number as the modulus. It is anticipated that the discrete logarithm problem with a composite number as the modulus is a difficult problem if the prime factors of the composite number, which is used as the modulus, are unknown. Then the problem can be applied to the cryptography. In the general discrete logarithm problem with a composite number as the modulus, it is not always true that an arbitrary element has a logarithm. From such a viewpoint, this paper shows that the exponent of an arbitrary element belonging to the irreducible residue class with a composite number as the modulus has a logarithm. Then the necessary condition in the application to the cryptographic technique is presented. Finally, as an application example of the technique proposed in this paper, a cryptographic technique based on the discrete logarithm problem with the composite number as the modulus is shown. [ABSTRACT FROM AUTHOR]

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

In the engineering and physical systems described by a lumped-constant system, sometimes a certain kind of symmetry is observed in the structure of the system or the nonlinear characteristics of the element. In such a system the symmetry is reflected on the state equation and the equation becomes invariant under the symmetrical transformation of the state variable. It may happen in the state equation with a symmetry that there exists an equilibrium point or a periodic solution, and the solution bifurcates into the solution without a symmetry when the system parameter changes. This phenomenon is called symmetry-breaking bifurcation. This paper discusses the symmetry concerning the equilibrium point and the periodic solution of the autonomous system and presents the condition for the symmetry-breaking bifurcation and the method to calculate the bifurcation parameters. It is pointed out that the symmetry-breaking bifurcation is, in general, a degenerate bifurcation and the bifurcation parameters cannot be determined by the ordinary method. It is shown that, by considering the symmetry, the calculation is made possible and is simplified. [ABSTRACT FROM AUTHOR]

This paper considers the case where hysteresis is given by the Preisach model and discusses the stability analysis for the periodic solution in the nonautonomous system containing hysteretic elements. The behavior of the system containing hysteretic elements is affected by the history of the system until it arrives at the steady state. In this paper, it is shown first that even periodic solutions with the same hysteretic loop may have different variational systems for the steady-state solution if the histories until arriving at the steady state are different, Next, two kinds of stability criteria are considered based on the difference of the history up to the steady state and the stability against the external disturbance is investigated. The dependence of the stability of the solution on the bifurcation of the periodic solution also is discussed. Finally, examples of numerical analysis are presented to verify the result of discussions. [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]

This paper summarizes the research result up to the present concerning the generic framework FITS for ITs. FITS is composed of six building blocks, which are teaching-material-independent, and the scheduler to realize the flexible teaching process. By developing the generic framework reflecting the essential task structure of ITS, the manpower needed in developing ITS can be reduced. This paper describes the design philosophy of the generic framework FITS and the mechanisms of the building blocks using the case of the teaching of geography as an example. [ABSTRACT FROM AUTHOR]

This paper discusses the system for automatic design of the parallel/pipelined VLSI architecture based on the signal processing algorithm. A new design language is introduced into the proposed system. The language is defined so that the time variable used in the signal processing algorithm can easily be described. The system has a high modularity, and the modules can be combined into a hierarchical structure. By this scheme, the algorithm with a high complexity can easily be described. The algorithm inputted to the system may contain a backward time-flow of the delayed data as well as a complex hierarchical. structure, which makes it difficult to execute the data-flow graph obtained directly from the algorithm by the parallel/pipelined structure. From such a viewpoint, this paper adopts the following procedure. The data-flow graph is derived from the given algorithm. A sufficient parallelism and a highly efficient pipelined mechanism are extracted from the obtained graph. Then the parallel processing architecture is designed. As an example, a parallel/pipelined architecture is designed for a simple adaptive signal processing algorithm. The result is compared with the traditional design examples, and it is shown that the efficient data-flow architecture is derived. [ABSTRACT FROM AUTHOR]

This paper considers the merging section of an APM (automated people mover), which is an approach to the new traffic system. The behavior of the vehicles at the merging section is modeled, and the situation, when different strategies are adopted, is analyzed using the technique of Markov chain. An APM system is considered, where the vehicles in the system do not always run individually, but sometimes run in the form of a train. This paper attempts to point out the essential dependencies of the vehicle behavior on the control strategy. It is assumed that the vehicles arrive individually. The basic idea of the control strategy is that the individually incoming vehicles are connected at the merging section. Thus the vehicle capacity of the outgoing line from the merging section is equivalently increased. As the result of analysis, the relations among the arriving traffic of the vehicles in two incoming tines, the average waiting time of the vehicles in the individual lines, the average number of connected vehicles in a train, and the density of train operations are derived explicitly. The dependencies of the average waiting time of the vehicle and the outflow traffic density of the train on different control strategies are examined. [ABSTRACT FROM AUTHOR]

If the signal space matched to the representation is selected according to the properties of the signal to be handled, the signal processing can be made efficient. this paper is a step toward the structuralization of the signal spaces for systematic selection of the signal space. As a particular problem, the relation is analyzed between the signal spaces composed of spline functions, which have the feature that the smoothness as the continuous differentiability varies according to the degree, and the band-limited signal spaces, which is fundamental in the signal processing. As a result of analysis, it. is shown that the signal spaces composed of spline functions agree with the band-limited signal spaces in the limit of infinite order. This implies that the signal spaces composed of spline functions and the band-limited signal spaces can be structured as a sequence with the degree as the parameter. The result in this paper provides a view that the continuous differentiability can be used as an index in the systematic selection of the signal spaces. [ABSTRACT FROM AUTHOR]

This paper presents an improvement of the Rabin cipher as a public key cipher for electronic authentication. The Rabin cipher has the merit of speed of processing and the clear specification of the safety condition. On the other hand, it has a problem in that four deciphered messages correspond to a ciphered message, and a complex verification procedure is required to check whether or not the digital. signature is correct. From such a viewpoint, this paper considers the transformation ∫(M)-g(A, M)·(A ‖ M) of the signatured original message M by suitably defining the function g and the bit sequence A. By this scheme, the transformation of the digital signature √∫(M) (mod p · q)&rarrM can be executed with one-to-one correspondence. Thus, it is made possible to decipher uniquely the digital signature within the specified computational complexity without adding extra information other than the digital signature of one block length. As a result, the digital signature based on the Rabin cipher is made more practical.

This paper describes a method of designing the fan filter with a steep cutoff characteristic, using McClellan transformation. The conventional design by McClellan transformation has a problem in that the cutoff region of the fan filter is broad due to the restriction to a lower order of the sub filter employed. From such a viewpoint, this paper proposes a systematic design method For the sub-filter with arbitrary order. First, it is pointed out that the ideal response ot the sub filter can be represented in a functional form. Then the error function is defined in regard to the idea] response, and the design is shown for the sub filter where the filter coefficients minimize the error function. Finally, a design example is discussed, and it is shown that a fan filter with a steeper cutoff characteristic than in the conventional method can be designed by the proposed method. [ABSTRACT FROM AUTHOR]

Recently, for signal processing such as echo cancellation and equalization it is required to develop IIR-adaptive digital filters. This paper proposes a new-type IIR- adaptive algorithm based on the orthogonalization method, considers the convergence of the algorithm, and verifies its effectiveness by simulation. White's algorithm is one of IIR-adaptive algorithms reported thus far and Feintuch's algorithm is known as its approximation algorithm. This paper also derives an IIR-adaptive algorithm using both Feintuch's algorithm and the orthogonalization method, First, it can be shown that a generalized algorithm (IIR-LI) of IIR-learning identification method based on previous heuristic methods can be derived using the idea of Feintuch's algorithm and the conjugate gradient method. Next, a new IIR-adaptive algorithm (IIR-AACG) based on the orthogonalization principle of the conjugate gradient method is presented. Further, by applying the orthogonalization method to a past input vector sequence we derive an IIR-Affine projection algorithm (1IR-AP). Finally, these algorithms are compared by computer simulation to verify the effectiveness of the orthogonalization method. [ABSTRACT FROM AUTHOR]

gggggAssume that an n × n dot matrix is stored in consecutive n words of a computer with one word being n bit. This paper discusses eight operations to rotate or transpose the dot matrix in horizontal/vertical! slant direction. As is well known, such operations form a group. This paper presents a unified description of the algorithms, and it is shown that all eight algorithms can be derived by suit- ably setting the constants in the unified algorithm. Then it is shown that the algorithms can be transformed to each other by simple transformation rules for the constants, and those transformation rules are shown to form a group. The foregoing algorithms for the dot matrix are not unique, and there can be different descriptions for the algorithm producing the same result. Consequently, there can be various versions of the eight algorithms as well as versions of transformation rules. This paper shows that one of the two groups formed by the transformation rules is isomorphic to the group of algorithms, but the other is not isomorphic. [ABSTRACT FROM AUTHOR]

gProgress in decreasing the power consumption of communications equipment has created a need for corresponding decreases in the power consumption of switched capacitor filters (SCF). Ishikawa and Laker haver suggested reducing the number of opamps used by means of time sharing, in the present paper a general form of a biquad SCF with a single opamp for time-sharing use is proposed and a new biquad SCF design is derived. The possibility of designing such SCF's with a small DC offset is demonstrated. Tests with individual components confirm the effectiveness of such devices. [ABSTRACT FROM AUTHOR]

One method used in the reliability design of a switching system is the determination of the unavailability target according to the scale of a failure. This method has already been applied to the design of digital switching system. The basic idea of this approach is the following. The effect produced by a failure in a switching system depends on the scale of the failure. Consequently, in order to achieve a balance between economy and reliability, the target of reliability should be set according to the scale of the failure. A similar idea can be applied to a link in a communications network. This paper presents the basic concepts involved in setting the unavailability target in the link of a communications network based on the scale of the failure. The effect of a link failure is formulated in terms of the degradation of the channel capacity, and is introduced as the network loss. A method is presented, wherein the unavailability target is set so that the loss is a reasonable value. In contrast to the case of a switching system, for a network it is a difficult problem to determine the unavailability target of a link. This is because a link is often a common path between different node pairs, and a strict reliability design should consider all the paths between nodes. This paper presents a simple method for dealing with this problem, in which the unavailability target is set taking into account only the scale of the link, and not the individual paths between the nodes. It is shown that the proposed reliability design method is quite useful in practice. [ABSTRACT FROM AUTHOR]

This paper proposes a new construction method for 2-D IIR digital filters with all-pass subfilters as components. Recently, numerous studies have been made to construct digital filters by combining all-pass characteristics. This paper is based on the construction by a parallel connection of an all- pass subfilter and delay element, which can realize approximately the linear phase characteristic. First, the basic design for 1-D digital filters as well as its properties are described. Then the theory for the 1-D case is extended to the 2-D case, and a new construction method for 2-D digital filters with low-pass characteristics with circular symmetry is shown. The proposed method features the following. The phase characteristic is approximately linear in the passband. The filter can be realized with a smaller number of multipliers than in the traditional method. The coefficient sensitivity is low. [ABSTRACT FROM AUTHOR]

Problems at present in the manual layout design of logic LSI are the high design cost due to the overhead of detection and correction of design errors, as well as the difficulty in modifying and reusing the design data. To solve these problems, this paper proposes a data representation providing the correspondence between the circuit structure and the layout, together with a new layout design method using that idea. This data representation leads to a unified treatment based on the correspondence between the circuit data and the layout data. It is made easy to modify the design and to maintain or reuse the design data. The layout design method based on the proposed idea effectively utilizes the circuit data in the layout design, which can completely prevent routing errors, and simplifies various layout verifications. Further, this paper presents an example wherein the proposed layout design method is applied to the symbolic layout system for MOS logic LSI. The result of layout experiments indicates the usefulness of the proposed design method. [ABSTRACT FROM AUTHOR]

In the analysis of a discrete nonlinear system, one of the important problems is to derive a steady solution for the nonlinear difference equation characterizing the system. However, the study for the solution method of the difference system has not been sufficient, and examples of the methods being applied to the parametric oscillation are few. This paper presents a method which derives the steady solution for the quasi-linear difference equation containing a periodic parametric excitation in its weakly nonlinear function with small parameter ∊. In this method the harmonic balance and the averaging methods used in the solution of the continuous nonlinear differential equation are ape plied to the discrete system. Since the non-linear term in the equation is bounded by ∊, the discrete waveform in the steady state is almost sinusoidal. This paper presents numerical examples for the solutions by the two methods, assuming the solution as the discrete oscillation corresponding to the sinusoid of the continuous system. As a result, the same solution is obtained by either method. Comparing the result with the output waveform obtained directly by the numerical calculation for the fundamental equation, the validity of the approximate solution by the proposed method was verified. [ABSTRACT FROM AUTHOR]

This paper presents a realization of an arbitrary conductance matrix using operational amplifiers, together with same applications. It is shown first that any conductance matrix can be realized using operational amplifiers, and the design algorithm which is convenient in the realization is given in the form of a flowchart. Then as an application, a construction of a practical matched bilateral amplifier is presented, together with its application. The realization of a gyrator is also shown. The method presented in this paper is a new and simple way to realize an arbitrary conductance matrix. The matched bilateral amplifier obtained by the proposed method is simple and better than the traditional circuit in terms of the number of circuit elements, element sensitivity, adjustment of the matching impedance, dynamic range and stability. As a practical example of application, the circuit is applied to the four-line carrier telephone link. [ABSTRACT FROM AUTHOR]

This paper proposes a first-order algorithm for unconstrained nonlinear optimal control problems based on the fixed point theorem. The method has the following features: (a) the algorithm is simple; (b) unlike second-order algorithms, the choice of the first approximation is less critical; and (c) unlike gradient methods, experimental selections of parameters during iterative calculations are not required. This paper also describes the conditions of convergence of the algorithm in the space of continuous functions, and the optimality of a solution derived. [ABSTRACT FROM AUTHOR]

Several methods have been proposed to determine a high-resolution NNR spectrum from limited data. This paper attempts to improve and extend the NMR spectrum estimation method using a complex Kalman filter, which was pro- posed previously by Nishlyama and Mita. In the proposed NMR spectrum estimation using the Kalman filter, extra data were required in addition to that to be used in the spectrum estimation, in the approximate determination of the spin-spin relaxation time. Hence, the previously reported NMR spectrum estimation is not based strictly on the same condition as that of the discrete Fourier transform. This paper improves this point, and the extra data are obtained by extrapolation by the AR model. This makes it possible to compare the method with the discrete Fourier transform based on the same number of data. It is shown that when the spectrum is limited to a certain frequency band, the proposed method gives an NMR spectrum with twice higher frequency resolution by reducing the effect of the spin-spin relaxation time and the effect of discrete approximation. In the proposed method, the spin-spin relaxation time of the nucleus, corresponding to the frequency resolution that is twice higher than usual, is determined by a linear interpolation of the spin-spin (relaxation time which is obtained approximately by discrete Fourier transform. [ABSTRACT FROM AUTHOR]

Several basic results on nonlinear circuit problems are presented from the topological viewpoint. There have been numerous papers which studied the relations between the boundedness of the solutions and the characteristics of the nonlinear resistors. An interesting question related to this was raised by Smale and Matsumoto. Smale assumed that there exists a compact set of attraction if the power of the resistors satisfy the eventual passivities. In addition, Matsumoto conjectured that the total number of singular points is odd if eventually the circuit is passive. This paper shows that the two forementioned conjectures are true under some additional conditions concerning the singular points of the circuits. For the proof of those conjectures, we used the Morse-Smale inequalities and the technique of the Gnomon projection (Poincaré compactification) which projects the n-dimensional Euclidean space onto the n-dimensional compact sphere Sn. [ABSTRACT FROM AUTHOR]

This paper presents a unified theory for the nonuniform sampling theorem for the low-pass band-limited waveform. In most practical cases, the mean interval of the sampling points is in the so-called oversampling state. One can also assume that there exists a certain period in the sampling point sequence. Those properties of the sampling point sequence are assumed in this paper. Yen has derived a sampling theorem for the case where the sampling-point sequence has a period. In his theorem, however, the interpolation function has a spectrum which is a stepwise discontinuous function, not suited to the realization of the interpolation function by a digital or other similar filters. One of the greatest features of the sampling theorem proposed in this paper is that the interpolation function has a continuous spectrum. The problem has been discussed by one of the authors for the case where the sampling frequency and the band-end frequency of the waveform are in an integer ratio. In this paper, the sampling theorem is derived in a unified and integrated form without imposing such a condition. The variance of the approximation error is also derived for the case where a statistically independent additive error is superposed on the sampled value, and it is shown that the proposed sampling theorem is better in that respect. The discussion is extended to the multidimensional band-limited waveform, and a sampling theorem is shown which has an interpolation function with a continuous spectrum. [ABSTRACT FROM AUTHOR]

This paper studies the analysis and minimization of roundoff noise in separable denominator multidimensional (SD M-D) digital filters on the basis of state-space representations. First, the roundoff noise in SD M-D digital filters is analyzed and the !2-norm dynamic range constraint is discussed. Using the results, a synthesis method of optimal realizations (with respect to roundoff noise) is proposed. Further, the simple relation between balanced realizations and optimal realizations of SD M-D digital filters is obtained and an efficient method for synthesizing optimal realizations from balanced realizations is proposed. In addition, this paper also proves that optimal realizations are free of overflow oscillations under zero input conditions. Finally, the synthesis procedure is illustrated by a numerical example. [ABSTRACT FROM AUTHOR]

It is known that a leapfrog-type digital filter obtained by simulating an LC ladder filter terminated with resistors on both ends is a digital filter with a low coefficient sensitivity. In this paper, an analog circuit to be simulated is chosen as a distributed circuit in which the real frequency and the discrete variable correspond rigorously. By means of an equivalent transformation, a leapfrog-type digital filter is constructed in which no delay free loop is generated. In this method, an LC ladder circuit expressed in terms of the Richard variable p is transformed to the circuit expressed in terms of the WI variable y. The digital filter is configured by stimulation of the relationship between the node voltage and series branch current in each element of LC ladder circuit. However, if the configuration is based on general distributed LC ladder circuits, a delay-free loop is always generated and configuration becomes unstable. As a solution to this problem, unit elements are introduced in this paper. It is shown that a realizable signal-flow graph can always be obtained if stub circuits are used as the original circuit. The basic section necessary to circuit configuration is presented. Finally, as a practical example, the present method is used for design of a fourth-order bandpass filter. The results are described and compared with those by other configurations. [ABSTRACT FROM AUTHOR]

Recently, the signal processing by the three-dimensional digital filter is considered in such applications as dynamic image processing. medical image processing, robot vision, and analysis of seismic or meteorological data. This paper proposes a method of optimum design for the three-dimensional FIR digital filter using linear programming. The optimum design of the three-dimensional digital filter, including the design by linear programming, requires in general a large computational complexity. From such a viewpoint, this paper discusses the reduction of computational complexity utilizing the symmetry in the frequency response, and the result is actually applied to the design by linear programming. As a result, it is shown, for the case of the spherical low- pass filter, that the computation time is reduced by one order of magnitude by utilizing the symmetry. [ABSTRACT FROM AUTHOR]

In the use of the three-dimensional data obtained from a stereo system, laser range finder, and other equipment, merge processing is important. This paper proposes a new merge method for range images, based on the volume-based technique. In the proposed method, in contrast to the conventional method, the voxels are adaptively segmented according to the curvature of the surface shape to be reconstructed. This helps to represent the geometrical shape efficiently, effectively utilizing the computation resources. In the framework of the proposed merge process, additional features such as color and the laser reflectivity can be added to the three-dimensional geometrical information, which helps to maintain sharp edges in the texture and to perform rendering and the texture mapping effectively. By examining the consensus for both the geometrical shape and the color in the whole framework, the merge is made robust to noise. In this paper, the system is described first, and then the results of application to several kinds of actual data are presented. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(11): 50–60, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20353 [ABSTRACT FROM AUTHOR]

In sensor networks, many of the sensor nodes are often positioned in clusters, and there is a correlation between the data obtained at the various sensor nodes by observation. This kind of observational data is collected at fusion centers. In this paper, we study the observation precision at fusion centers from the viewpoint of the Kullback–Leibler (KL) information. In particular, we consider transmission path errors arising when transmission is made from each sensor node via wireless communication paths, as a factor in the deterioration of the KL information, or in other words, observation precision, and study its effects and methods for reducing it. In this paper, we study the KL information obtained in fusion centers when there have been transmission path errors in cases where the correlation between observational data is all equal, and where the correlation is random. We show that the deterioration of the KL information can be reduced in fusion centers by using the correlation of observational data in the estimates of error data. In addition, we present examples of estimating the correlation between observational data in fusion centers that use correlation, and of estimating error data using correlation. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(11): 1–10, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20250 [ABSTRACT FROM AUTHOR]

This paper discusses a method for increasing the eigenmode SNR by using a space-time linear precoder designed on a maximum likelihood (ML) basis (henceforth called the space-time linear ML precoder). First, the effect of the length of the precoded signal generated in the ML precoder on the eigenmode SNR is studied. It is shown that the eigenmode SNR can be increased by varying the precoded signal length depending on the condition of the channel. Next, a transceiver configuration for increasing the eigenmode SNR by the space-time linear ML precoder is presented. In the proposed method, error propagation may be generated due to the transceiver configuration and the BER characteristics may be degraded, even if the eigenmode SNR is increased. In this paper, in order to reduce the effect of the error propagation on the BER characteristics, the relationship between the average length of the error propagation and the precoded signal length in the proposed configuration is presented. It is demonstrated that the effect of error propagation can be minimized by limiting the region of search for the precoded signal length. Finally, the effectiveness of the proposed method is verified by numerical simulation. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(10): 1– 8, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20304 [ABSTRACT FROM AUTHOR]

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