bluuebunny/crossref_metadata_2025_split huggingface dataset:
Elastic and piezoelectric properties of piezoelectric composite materials
Authors: Hashimoto Ken‐Ya , Yamaguchi Masatune 
Published: July 2007

AbstractThis paper offers theoretical and experimental examinations of the elastic and piezoelectric properties of piezoelectric composite materials used mainly for ultrasonic medical‐diagnosis apparatus. One‐dimensional multilayer composite materials in which the thickness of each layer is less than the wavelength of elastic waves (i.e., each layer can be regarded as uniform), were analyzed rigorously, and their equivalent elastic, piezoelectric, and dielectric constants were obtained. By using the results, the characteristics of a vibrator employing the one‐dimensional composite‐material (piezoelectric ceramic and epoxy resin) were examined. the relationship between the high‐coupling and low‐impedance (which are the features of this type of composite material) and the behavior of each component material was investigated. A model which evaluates the characteristics of the 1‐3 type twodimensional composite piezoelectric material (which is important for practical ultrasonic transducers) was proposed. the characteristics of a vibrator made of this material were examined. Not only can this model explain the known experimental facts, but it can easily deduce the piezoelectric constants which are important for evaluating the spurious response. A vibrator using the composite material was fabricated. Its characteristics agree reasonably well with the result of the analysis. the proposed method can be expanded for composite materials other than the 1‐3 type, and will be useful for the design and development of general compound materials.
Instruments for displaying the Lorenz maps and the Poincaré maps on a synchroscope
Authors: Itoh Makoto , Tomiyasu Ryuichi 
Published: July 2007

AbstractThis paper outlines the new instruments for displaying Lorenz maps and Poincaré maps on a synchroscope. These instruments are designed using only a few ICs, and, as a result, they are easily built. Next, we present the observation examples of the forementioned two maps obtained from the chaotic circuit. Furthermore, it is shown that the shape of the maps depends not on the nature of uncertain factors, but on the structure of the bundle of the solutions.
Linear phase FIR digital filters using interpolated equalizers
Authors: Honma Hitoshi , Yamada Masakazu , Sagawa Masahiko 
Published: July 2007

AbstractA method for designing an FIR digital filter is reported with a view to reducing the number of multipliers and addres. an effective method for reducing the number of multipliers and addres is the one in which a prefilter without a multiplier and an equalizer (compensating the characteristics to satisfy the given specifications) are cascaded. In previous studies on this method, the design of the prefilter was the main topic and no sufficient attention was given to the equalizer. In the present method, the design method of the equalizer is also studied. A new design method for an equalizer is proposed in which a concept of interpolation is introduced. the present method is applied to a number of examples. It is confirmed that a significant reudtion of the number of multipliers and addres is obtained.
Normalized ARMA lattice filter and its terminal conditions
Authors: Haseyama Miki , Nagai Nobuo , Miki Nobuhiro , Miyanaga Yoshikazu 
Published: July 2007

AbstractIn this paper it is shown that covariant data of a synthesized signal corresponds to the covariant data of a real signal.This occurs when an ARMA lattice synthesis filter is used and the minimum mean square value of the estimation error is zero.Hence, a new terminal condition of the ARMA lattice filter is derived so that the covariant data of the synthesized signal will always correspond to the covariant data of a real signal.To this end, a design method for the normalized ARMA lattice filter is presented in this paper, and normalized estimation errors are derived. Finally, the starting condition and design for the normalized ARMA lattice filter is given.
Universal learning machine and application to english type‐letter recognition
Authors: Suzuki Hisashi , Arimoto Suguru 
Published: July 2007

AbstractThis paper proposes a universal learning machine (ULM) that absorbs information from environments and acquires an ability to process information professionally according to the environments. Additionally, it applies this information to the recognition of the English type‐letters that are highdensely printed in magazines, books, etc. A window is applied on a black and white image gathered through an image scanner so that only a letter and a head part of the succeeding letter can be observed. Appropriately fixing a distance that measures the similarity between the window images, the recognizer utilizing the ULM searches for some one of the most similar window images to the current from among the window images memorized in the past. Then the recognizer loads the records accompanying it to display for a user the reading of the designated letter and the boundary between this letter and the succeeding letter. the user commands whether the recognizer should revise the memory or continue the recognition process. If the command is for revision, the content of revision is memorized in the form of a binary tree while part of the content memorized in the past is deleted to prevent overflow in system memory capacity. A theoretical evaluation of the performance of the ULM and an experimental evaluation on the English type‐letter recognizer actually constructed are performed.
Frequency spectrum estimation method using complex Kalman filter
Authors: Nishiyama Kiyoshi , Mita Tutomu 
Published: July 2007

AbstractAs a method to estimate the frequency spectrum, the discrete Fourier transform is employed widely, since a high‐speed algorithm is available. However, the discrete Fourier transform has the following problems. the presence of noise is not considered, and the resolution of the frequency spectrum is determined uniquely by the sample duration. Other problems are that the sampling interval must be kept constant and the decay of the signal is not considered. This paper proposes a high‐resolution frequency spectrum estimation where the complex Kalman filter is applied to the observed signal represented by the complex stochastic system, effectively utilizing the a priori knowledge of the signal source. the method is applied to the measured data and to the simulation, and it is demonstrated that a high‐resolution frequency spectrum is obtained by suppressing the effect of the discrete approximation of the frequency component, and, when the frequency spectrum is concentrated, the resolution can be improved approximately by a factor of two, without iterating the measurement. It is shown also that the sampling interval can be set arbitrarily in the proposed method, and the noise and the decay can be considered. A theoretical discussion is made on the relation between the special case of the proposed method and the discrete Fourier transform.
Stability domain of the second‐order discrete oscillatory system with parametric excitation
Authors: Tanaka Toshiyuki , Sato Chikara 
Published: July 2007

AbstractThis paper presents a quantitative discussion on the stability of the second‐order periodic difference equation, which characterizes the discrete periodic time‐varying system. the periodic parameter discrete system, which corresponds to Mathieu's equation in the continuous system, is represented by a second‐order linear difference equation with a small parameter ε in the varying term. the parameter ε plays an important role in the determination of the stability. By applying McLachlan's method, the expression for the boundary between the stability and the instability can be determined analytically with regard to the parameters contained in the equation. For the case where the periodic parameter of the difference equation is represented as a sum of two even functions, the boundary between stability and instability is determined. the stability can be analyzed in a similar way for the case where the periodic parameter is represented as a sum of N even functions in Fourier series.
Updating of all the shortest paths
Authors: Izumi Takashi , Takahashi Yutaka , Kawanishi Kenji 
Published: July 2007

AbstractThere are times when updating of shortest paths is required when a part of a network structure, such as length of a branch, is altered. For updating shortest paths, information on already obtained shortest paths including mid‐progresses can be used. Many of the algorithms shown so far are aimed at altering the length of one branch in a network or some branches going in or out at a node. With these algorithms, it is necessary to repeat the application if there are simultaneous alterations of multiple branches other than described in the foregoing, and the efficiency is not always good in terms of computation steps. This paper describes an algorithm which updates shortest paths between all pairs of nodes when simultaneous alterations of lengths of multiple branches, with no restriction to branches going in/out at one node, are allowed. In terms of computation steps, our algorithm is more efficient than repeatedly applying conventional methods.
An efficient probabilistic algorithm for solving quadratic equation over finite fields
Authors: Itoh Toshiya 
Published: July 2007

AbstractThe probabilistic algorithm has a random procedure, which may or may not produce the solution, due to its randomness. By iterating the application, the desired solution can be produced with a high probability (unless the probability that the probabilistic algorithm produces the solution is very low). In 1980, Rabin proposed a probabilistic algorithm to solve a higher‐order equation over a finite field. When his algorithm is applied to the quadratic equation on a finite field, the probability that the desired solution is obtained by one application is 1/2, and the root can be determined by two times applications on the average. This paper proposes a probabilistic algorithm in which the probability that the desired solution is obtained by one application is kept as one‐half, and the quadratic equation over the finite field GF(p) (where p is an odd prime number) or GF(2m) is solved more efficiently than by Rabin's method. the proposed algorithm calculates the solution more efficiently than Rabin's method by determining the random parameter to be set in finding the solution, using a procedure which can efficiently be executed before calculation.
Correction theory and algorithm for reachability matrix model
Authors: Kase Satoshi , Ohuchi Azuma , Kaji Ikuo 
Published: July 2007

AbstractA reachability matrix M is a binary matrix with the reflexive and transitive property, i.e., M + I = M and M2 = M, where I is the identity matrix. Reachability matrices are used to identify structural models of complex systems. While a structural model is being developed or after it has been developed, the developer may want to make changes in it. Procedures for changing a structural model with computer assistance are called “correction procedures.” Several algorithms for correction procedures of the partially filled reachability matrix are de scribed.
Monotonic properties for the responses of a class of CMOS circuits
Authors: Urahama Kiichi 
Published: July 2007

AbstractThis paper considers a CMOS circuit which is composed of inverter‐type gates, transfer gates, and connections. the properties of the equivalent circuit obtained by replacing those elements by circuit models are discussed. It is shown that if all inverter‐type gates in the circuit can be given signs so that different signs are adjacent to each other, the response waveform of the circuit depends monotonically on the initial value, input and the transistor parameters.
Estimation of the frquency of a sinusoidal wave signal with short record length
Authors: Konishi RyōSuke , Matsumoto Shigeharu , Sasakura Hiroshi 
Published: July 2007

AbstractThis paper presents a new method of an accurate estimation of a sinusoidal wave signal with short record‐length and with noise. the method is based on the linear prediction model (L.P. model). Although a conventional L.P. model is known to be an accurate method for estimating a sinusoidal wave with a short record‐length, the authors found that even minor noise degrades its accuracy significantly. to improve this noise dependence, the authors introduced two new algorithms: (1) restriction of the coefficient value in the linear prediction equation; and (2) averaging of frequencies which are obtained from the time delay τ around a point where the sign of the coefficient reverses. the results obtained by this method show that the frequency‐estimation error is reduced to one‐fifth to one‐ tenth that obtained by a conventional L.P. model when the record length corresponds ‐to 1.2 period of a sinusoidal wave with noise, confirming the effect of the proposed method.
On the LP‐HP transformation in SCF
Authors: Ishikawa Masayuki , Fujii Nobuo 
Published: July 2007

AbstractTo realize a highpass filter structure from a lowpass one, one can use the lowpass‐ highpass transformation (LP‐HP transformation) technique. This technique can easily be applied to passive LC filters, active RC filters, and digital filters. However, the direct application of the LP‐HP transformation to SCF's has not been reported. This paper shows that the LP‐HP transformation by replacing z with ‐z as is used in digital filters, is applicable to SCF's. We can easily obtain highpass characteristics from a lowpass SCF by adding a simple transforming circuit to the lowpass SCF. In addition, the HPF obtained from its LPF structure can have low element sensitivities and a narrow element value range.
Cluster‐based management of large computer networks. Computational complexity of initial clustering problem
Authors: Ishida Kenji , Miyao Jun'Ichi , Kikuno Tohru , Yoshida Noriyoshi 
Published: July 2007

AbstractOne of the interesting issues in the management of large computer networks is to partition the network into clusters and then to perform the management for each cluster. This paper presents a basic discussion on the initial clustering problem for a given network. the initial clustering problem C is defined as follows. Given a graphic representation G=(V, E), [V] = n of a network and a set of nodes C=[ci]1≤ i ≤ m ≤ V, the graph G is to be partitioned into m connected subgraphs Gi = (V≤ E≤) ( 1 ≤ i ≤ m) of G satisfying the following conditions: ≤ V, ≤ (a positive integer); and ≤ the distance between ≤ Vi and ≤ is minimum when × = i. the following results were obtained for the time complexity of the problem C: (1) the initial clustering problem C is in general NP‐hard; and (2) the sub‐problem obtained by deleting the condition ≤ can be solved in O (n2).
Design of multirate FIR digital filters with arbitrary passband
Authors: Hirai Ryoji , Ikehara Masaaki , Takahashi Shinichi 
Published: July 2007

AbstractThis paper proposes a new method of con structing the multirate FIR digital filter with arbitrary passband, using real‐coefficient filters for the decimator and interpolator composing the multirate filter. It is already known that the bandpass filter can easily be realized using the complex‐coefficient filter. In such a construction, however, the processing of the analytic signal is required using Hubert transform, which makes the whole system somewhat complex. From such a viewpoint, this paper pre sents a method of constructing the multirate bandpass digital filter using the real‐coefficient filter. Another method is also presented, where the muitirate lowpass filter with arbitrary passband can be constructed by combining the complementary response and a lowpass filter. the multirate method is effective in realizing a narrowband response, but may not be useful in realizing a relatively wideband response. By the proposed method, the latter can easily be realized, drastically reducing the number of multiplications. Similar methods can be applied to highpass and bandstop filters.
A high‐speed parallel FFT‐processor for a radio astronomy telescope
Authors: Iwase Seiichiro , Okita Toshimichi , Yamazaki Takao , Daishido Tsuneaki , Asuma Kuniyuki 
Published: July 2007

AbstractA radio astronomy telescope with a wide field of view is an efficient tool for locating such transient radio sources in space as supernova explosions. A promising method of designing this type of telescope is to apply Fourier transformation to the output signals from a phased array antenna. the implementation of a high‐speed parallel FFT processor to an eight‐element linear phased array antenna output is reported in this paper. Small size and low‐power con sumption are achieved by utilizing conventional digital video signal processing LSIs. In the present pilot system, arithmetic operation up to a speed of 2.7 COPS (giga operations per second) has been attained. the basic operation has been confirmed by solar observation.
Compression of digital images based on vector quantization with variable block size
Authors: Yahagi Takashi , Suga Akira , Takamura Matsuzo 
Published: July 2007

AbstractThe vector quantization recently has been considered interesting as a highly efficient coding for compressing digital images. For this purpose, the predictive coding and the transform coding have been known, but the vector quantization can realize the coding with fewer bits per pixel, which is a drastic compression compared with past methods. the image contains parts with different local properties such as parts with rapid intensity changes such as edge, and parts with slow intensity changes such as flat part. If this situation is taken into consideration, a more efficient coding can be realized. In the usual vector quantization, the overall image is divided into blocks of a constant size. In the coding based on the local property of the image, however, it is desirable to use blocks with variable sizes rather than a constant size. From such a viewpoint, this paper proposes a new vector quantization in which the block size resulting from the division is adjusted adaptively according to the local property of the image. First, the vector quantization with variable block size is described in detail. Then the proposed method is applied in several simulation problems, and the actual usefulness of the method using variable block sizes is demonstrated.
Deconvolution filtering for the high‐resolution ultrasonic pulse‐echo measurement
Authors: Yamada Akira 
Published: July 2007

AbstractIn the ultrasonic pulse‐echo measurement system, the deconvolution signal processing enables maximum information to be extracted from the echo signal under the physical restriction of the acoustic propagation loss, thereby realizing significant improvement of the resolution capability. This paper presents the deconvolution filtering technique which satisfies the practical requirements for the mitigation of noise amplification compatible with a simple fast procedure. Specifically, the time‐domain deconvolution filter is determined in such a way that the second‐order moment of the desired response is minimized under the band‐limited condition, thereby optimizing the filter coefficient subject to the constraints of the small processing noise. In addition, the envelope detection is incorporated for the removal of the distortion due to the band‐limited nature of the signal. Furthermore, the theoretical formula for the range resolutions achievable with this technique is derived. At the end of this paper, results of the evaluation experiments are shown to demonstrate that high‐resolution measurement comparable to the theoretical limit is obtained with small processing noise.
Sampling theorems using nonuniform sample points and interpolation functions with band‐limited continuous spectra
Authors: Kida Takuro , Akagawa Keiichi 
Published: July 2007

AbstractThis 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, how ever, 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.
Ultimate boundedness of uncertain linear systems under some control laws
Authors: Sugaya Jyunichi , Takeda Hiroshi 
Published: July 2007

AbstractThis paper investigates the ultimate boundedness of a linear system with a bounded uncertainty under some control input. These control inputs are determined so that the size of a bounded region, which ultimately the state of system reaches (the region of ultimate boundedness), is reduced. Then they are introduced as new control laws which are different from existing methods. These control laws consist of linear control laws for stabilizing a linear time‐invariant system, which is a basis for de signing, and modified control laws for making linear systems with disturbances, observation noises and uncertainty in system parameters, ultimately bounded. the control parameter contained in the latter modified control laws is determined so that the re gion of ultimate boundedness is reduced as much as possible by Lyapunov's second method. Control laws are linear when the control paprameter is a constant, and nonlinear when the parameter is a function of the state.In this paper, it is shown that the size of the region of ultimate boundedness obtained by the control laws newly proposed here is smaller than that obtained by the conventional Leitmann nonlinear control law.
Sampling bases for the signal space composed of spline functions
Authors: Kamada Masaru , Toraichi Kazuo , Mori Ryoichi 
Published: July 2007

AbstractThe 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. Exam ples 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.