bluuebunny/crossref_metadata_2025 huggingface dataset:
Parametric loudspeaker—characteristics of acoustic field and suitable modulation of carrier ultrasound
Authors: Aoki Kenichi , Kamakura Tomoo , Kumamoto Yoshiro
Published: July 2007

Due to the inherent nonlinearity of the medium, finite amplitude ultrasound interacts with itself and generates some secondary waves in the sound beam. The parametric loudspeaker, making use of this phenomenon, has a sharp directivity and might be applied to a speech transmission system under the limited environment.

In this paper, a numerical analysis method available for high sound pressure levels, where the nonlinear interactions of ultrasound become greatly active, can be used successfully to theoretically design the parametric loudspeaker. It is reported that the numerical computations agree well with the experiments by a circular aperture projector of radius 21 cm and carrier frequency 27 kHz.

To develop the parametric loudspeaker for practical uses, the problems on harmonic distortions and the physiological effect on human being must be solved. Based on the theoretical prediction, the reasonable solutions for such problems are considered in accordance with appropriate primary wave modulations.
Adapative formant estimation with compensation for gross spectral shape
Authors: Toyoshima Takuji , Miki Nobuhiro , Nagai Nobuo
Published: July 2007

Formant estimation from a speech signal that is observed acoustically is importand for speech analysis and synthesis. This paper proposes an adaptive formant estimation method with compensation for the gross spectral shape. This estimator can estimate the center frequencies and the bandwidths of formants (anti‐formants) based directly on the pole‐zero transfer function represented by the products of the secondand first‐order factors. Since the proposed method can compensate the gross spectral shape adaptively, excellent formant estimation is expected for continuous speech whose gross spectral shape is time‐varying. Therefore, the method proposed herein is applied to the synthetic signal and real speech, and the validity of this method is demonstrated. Furthermore, the Single‐Stage Iteration Filter is the estimation algorithm introduced for use in the proposed method. It is shown that the estimation results of tracking with little degradation are obtained stably in the transient part of the spectra.
Nondeterminism and existence of finite‐state supervisors in discrete‐event systems
Authors: Ushio Toshimitsu
Published: July 2007

Supervisory control initiated by Ramadge and Wonham is a very useful control technique for discrete‐event systems. A supervisor realized by a finite automaton is called a finite state supervisor. It is clear that a finite state supervisor exists for every regular controllable language. It is very important to show a necessary and sufficient condition for the existence of a finite state supervisor. Moreover, there are few studies for the effect of nondeterminism in a supervisor on control.

First, this paper shows that there exists a deterministic finite state supervisor for every control specification for which a nondeterministic finite state supervisor exists. Then a necessary and sufficient condition for the existence of a finite state supervisor is proved. As a result, it is shown that no finite state supervisor exists if a controlled discrete‐event system is modeled by a finite automaton and control specification is given by a nonregular language. Finally, a simple manufacturing system and a nonregular control specification (for which a finite state supervisor exists) are demonstrated.
Hierarchical detailed floorplanning with global routing in VLSI layout design
Authors: Ohmura Michiroh , Wakabayashi Shin'Ichi , Miyao Jun'Ichi , Yoshida Noriyoshi
Published: July 2007

In a VLSI layout design using the building block approach, the design is divided into two phases, placement and routing. On the other hand, a new hierarchical floorplanning method was proposed by Dai et al., in which a global routing for the evaluation of the placement is determined simultaneously during the design of relative placement. However, the precise estimation of routability is difficult since the global routes in this method do not correspond to the routing region such as channels and switchboxes. In this paper, a new method is proposed, which simultaneously and hierarchically obtains floorplan, shapes and positions of modules, and global routes which directly correspond to switchboxes and channels. In this method, the routing‐based partitioning, the hierarchical detailed global routing, and the hierarchical positioning are repeatedly executed, and a chip with small area and short wire length is obtained. This paper presents these three algorithms, and discusses the experimental results of each algorithm and the whole proposed method in a comparison to the conventional method that separately executes placement and routing.
An algorithm for generating low autocorrelation‐sidelobe skew symmetric binary sequences based on complex exponential functions
Authors: Ohue Kenji , Aibara Tsunehiro
Published: July 2007

In the smearing process to suppress impulse noises and the pulse compression in long‐range radar, it is crucial to employ the finite binary sequence which minimizes the sidelobe of the autocorrelation function.

This paper proposes a new generation algorithm for the finite binary sequence with a sharp autocorrelation function. The method is based on the fact that the autocorrelation function θ(n) of skew‐symmetric sequence is always zero when n is odd, and the Fourier expansion coefficient sequence of the complex exponential function is an uncorrelated sequence.

First, the generating function is derived for the skew‐symmetric real‐valued sequence, for which the Fourier expansion coefficient sequence is an uncorrelated sequence. An algorithm is presented in detail, where the foregoing real‐valued sequence is quantized into a binary sequence, and the binary sequence minimizing the autocorrelation is derived with the finite delay coefficients of the generating function as the variables.

In the past, no generation algorithm has been found which can derive the optimal skew‐symmetric binary sequence. It is shown that the proposed generation algorithm can derive all of the optimal skew‐symmetric binary sequence derived by Golay using the exhaustive method. It is shown also that all Barker sequences with off length can be derived.
A method for numerical analysis of bifurcation of periodic solutions in nonautonomous systems with hysteretic elements
Authors: Matsuo Tetsuji , Kishima Akira
Published: July 2007

This paper proposes a method for a numerical analysis of the bifurcation in the nonautonomous system containing hysteretic elements. Previously, the authors discussed the case where the characteristics of the hysteretic elements can be represented by Preisach's model, and proposed a method for steady‐state analysis of the nonautonomous system containing hysteretic elements.

First, a method is presented to investigate the change of the steady‐state behavior with the variation of the system parameters by combining the proposed method and the continuation method. Then another method is presented in which the singular point on the characteristic curve (solution curve) is determined using the previously derived variational equations of the system. Finally, examples of analysis by the newly proposed method are presented, and the effects of the hysteresis loss and the ohmic loss on the bifurcation are compared. A remarkable difference between the two is indicated.
Influence of time constant on coefficients estimation error derived from a lowpass filter expression for learning identification algorithm
Authors: Fujii Kensaku , Sakai Yoshihiro , Ohga Juro
Published: July 2007

This paper shows that the learning identification (the normalized LMS) algorithm can be expressed by a first‐order IIR low‐pass filter having rapidly varying coefficients as a function of output signal to be sent to an unknown system to be estimated. Based on the expression, this paper studies the estimation error of adaptive filters obtained by the learning identification algorithm. It is pointed out that the mean of estimation error given by the conventional analysis corresponds to the output fluctuation which can be computed by replacing varying coefficients of the lowpass filter with its arithmetic mean. However, the output fluctuation is affected by the time constant specified by the coefficients of the lowpass filter and it has a characteristic that it increases rapidly when time constant is short and decreases slowly when it is long. This means that the actual estimation error becomes larger than the result given by the conventional analysis.

According to the analysis shown in this paper, it can be seen that the estimation error consists of elements for which the portion increased by the effect of time constant become fewer anti‐proportionally to the number of the tap, and those elements which increase at a constant rate independently of the number of the tap. It was shown also by simulation using white noise that the difference between the estimation error obtained in actual estimation operation which occurs when the number of the tap is small and the result given by the conventional analysis can be explained well by the analysis by simulation using white noise.
A design method of 3‐D fir digital filters detecting linear trajectory signals using the rotation and interpolation in the signal domain
Authors: Kondo Katsuya , Sugii Yasuyuki , Hamada Nozomu , Xue Pei Ding
Published: July 2007

This paper proposes a method for designing an FIR digital filter which extracts an object moving in a fixed direction at a fixed speed on a plane (linear trajectory signal) with some margins concerning direction and speed independently. Signals to be considered are 3‐D signals on the 2‐D space‐time coordinates.

The method herein enables processing by FIR‐type prototype filter of cubic‐symmetric characteristics by rotating signal data and designing a prototype filter in the digital domain directly. Since this design method is not based on optimization design such as an iterative algorithm but finds filter coefficients in an analytical manner, the computation time is extremely short. In addition, since it is a direct design in the digital domain, it does not require s‐z transformation such as bilinear transformation and thus it is not affected by distortion caused by it. Moreover, because of the FIR type its feature is that we do not need to consider a stability check or phase distortion.
A distance measure for speech recognition based on an FM‐neuron model
Authors: Aikawa Kiyoaki , Furui Sadaoki
Published: July 2007

This paper discusses the speech recognition based on the time course of the local peak of the spectrum such as the formant, which has been considered important in the phoneme perception. A measure for the dynamical behavior of the spectrum is proposed based on the functional model of the FM‐neuron which is shown to exist in auditory physiology.

First, the FM‐neuron is modeled as a time‐frequency filter for the spectral time‐series which responds only to the shift of the local peak frequency of the spectrum with the discrimination function for the shift direction. Then the measure to represent the difference of the output from the FM‐neuron model is derived based on the cepstral expansion of the spectrum. The measure is called the spectral movement similarity.

It is shown that the spectral movement similarity on the auditory nonlinear frequency axis can be realized equivalently by the frequency weighting. A spoken word recognition experiment is conducted employing the dynamic time warping (DTW) using the spectral movement similarity. It is shown also that the recognition error is reduced greatly by combining the proposed measure and the traditional spectral distance compared to the case where only the traditional spectral distance is used. This improvement is more remarkable when the cepstral distance is used as the spectral distance, with the recognition error being reduced to one‐fourth.
A periodically time‐varying digital filter containing a discrete fourier transformer
Authors: Min Xiong Wei , Ishii Rokuya
Published: July 2007

So far, three periodically time‐varying models have been proposed in the research on periodically time‐varying digital filters. They are the coefficients time‐varying model, the output sampling polyphase model, and the input sampling polyphase model. Analyses of these models have also been completed by the authors.

This paper proposes new models to solve a problem which the three conventional models cannot solve. It is known that a periodically time‐varying digital filter can be used to implement a spectrum scrambler. However, when these models are to be used to implement a spectrum scrambler, time‐invariant filters in these models are restricted to FIR filters.

This paper proposes and analyzes an output sampling polyphase model and an input sampling polyphase model, both of which contain a discrete Fourier transformer. Relationships between the newly proposed models and the conventional coefficients time‐varying model are established.

By making use of the newly proposed periodically time‐varying models, we can design a spectrum scrambler employing not only FIR time‐invariant filters but IIR time‐invariant filters as well. Furthermore, compared to the conventional models, the newly proposed periodically time‐varying models are shown to have an advantage of less amount of computation when they are used to implement a spectrum scrambler.
Optimum linear filter of PPM receiver
Authors: Inoue Tadaaki , Iritani Tadamitsu , Kinouchi Yohsuke
Published: July 2007

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.
Improving the efficiency of interval analysis by Kevorkian's decomposition technique
Authors: Yamamura Kiyotaka , Ushida Akio , Horiuchi Kazuo
Published: July 2007

The algorithm of interval analysis proposed by Krawczyk, Moore, and Jones (henceforth called the Krawczyk algorithm) is a well‐known algorithm for finding all solutions of nonlinear equations. In this algorithm, the initial region (which is given by ann‐dimensional rectangle) is divided into small subregions, and the existence of a solution in each subregion is examined. Therefore, it can find all solutions. However, the computation time of the Krawczyk algorithm grows exponentially with the dimension, hence it is not a practical algorithm for large‐scale problems although high‐speed computers are used.

In this paper, Kevorkian's method is introduced to the Krawczyk algorithm to improve the computational efficiency. Kevorkian's method is a decomposition method for nonlinear equations. It reduces large‐scale nonlinear equations with sparse Jacobians into smaller equations. By this reduction, the computational efficiency of the Krawczyk algorithm is improved substantially.

First, a new algorithm of interval analysis is proposed in which Kevorkian's method is introduced to the Krawczyk algorithm. Then a new permutation algorithm (algorithm for decomposing nonlinear equations) that makes the interval arithmetic well defined is proposed. The effectiveness of the proposed algorithm also is verified by numerical examples.
Realization of multirate digital filters with powers‐of‐two coefficients
Authors: Todaka Chiaki , Kambayashi Noriyoshi
Published: July 2007

With a view to reducing the amount of hardware in the lowpass filter (LPF), bandpass filter (BPF), and band‐elimination filter (BEF) processing important in digital signal processing, this paper presents a method for configuration of the multirate digital filters with powers‐of‐two coefficients only. The proposed filters are constructed with the sampling rate increaser, the reference LPF and the sampling rate decreaser. The reference LPF consists of the stopband‐forming filter and the pass‐band compensating filter. The stopband‐forming filter consists of a cascade of RRSs while the linear phase FIR or IIR configuration is used for the passband‐compensating filter.

First, a method of configuration is presented for the reference LPF based on the sampling rate increaser with powers‐of‐two coefficients. Next, it is shown that the multirate LPF, HPF (high‐pass filter), BPF and BEF with an arbitrary cutoff frequency can be constructed easily from the reference LPF. The present method is particularly effective for narrowband LPF and HPF. Since no multiplier is used, they have zero coefficient sensitivity. It is shown by means of a design example that the present method is effective in reducing the amount of hardware.
Analysis of periodic oscillations with computer algebra
Authors: Fujisaka Hisato , Sato Chikara
Published: July 2007

In analyzing differential equations, computer algebra or symbolic computation is useful when we want to know the relation between the solutions and the parameters at the point near bifurcation.

The object of this paper is to show the fast computation method by means of computer algebra in analyzing the bifurcation problem from the steady state to the periodic oscillatory state. The proposed method of analysis depends on Lyapunov‐Schmidt method. Instead of using trigonometric multiplication and sum formula in computation of nonlinear terms and their harmonics, both the basis vectors with polynomials in components and their inner products are utilized to reduce the computation time significantly.

The example of a second‐order Van der Pol equation and the example of the higher‐order coupled system are used to give the bifurcation equation. Thus, the relation between the amplitude of oscillation and the parameter is given with the stability condition. By the proposed method with the aid of a microcomputer and symbolic computation language of interpreter type, the computation time is shorter.
Effects of respiration on heart‐rate variability measures
Authors: Murata Atsuo
Published: July 2007

There are many studies that measure the mental workload or evaluate the autonomous nervous function by heart‐rate variability measures. Heart‐rate variability is influenced not only by the stress or the workload but also by respiration. Therefore, it is important to clarify the effects of respiration on heart‐rate variability measures.

This paper discusses the effects of the respiration frequency, the depth of respiration and the dispersion of the respiration interval on heart‐rate variability. It was shown that the number of peak and trough waves in the time series of R‐R intervals was influenced by the respiration frequency. Moreover, the measures obtained by the power spectral analysis of R‐R intervals were found to be influenced considerably by the respiration frequency. The heart‐rate variability measures tended to increase with deep respiration. The effects of the dispersion of respiration intervals on heart‐rate variability measures were not so remarkable.
Theory on extended form of interpolatory approximation of multidimensional waves
Authors: Kida Takuro , Yoshioka Leopoldo Rideki , Takahashi Sadayoshi , Kaneda Hajime
Published: July 2007

This paper presents a comprehensive discussion of the approximation of then‐dimensional wavef(X) using the sampled values of the output wave obtained by exciting a series of time‐invariant linear circuits by the wavef(X). It is assumed that the approximate waveh(X) off(X) is given by the sum of sample values of the output wave multiplied by certainn‐dimensional waves. For simplicity,n‐dimensional waves to be multiplied with the sample values are called the interpolation functions.

The set of sample points treated in this paper is defined as a subset obtained by sampling periodically the vertices of then‐dimensional parallelepipeds placed periodically in the spaceRn.Such a set of sampling points includes the most of the typical arrangements of the sampling points, such as the hexagonal and the octagonal lattices on the two‐dimensional space.

It is assumed that the sample values contain statistically independent errors such as the observation error and/or the quantization error. Moreover, it is assumed that the interpolation functions have the supports which are parallel‐translations of each other.

First, it is assumed that the functional forms of these interpolation functions may be different. Further, a set ofn‐dimensional waves is considered where the corresponding spectrum has the weightedp‐norms smaller than the prescribed positive constant. The standard deviations of the difference betweenf(X) and their approximations are considered. As the measure of the approximation error, the upper limit of the standard deviation obtained by varying the original waves over the given set of waves is adopted.

In the following sections it is shown that the interpolation functions minimizing the forementioned measure of error can be expressed as the parallel‐translations of a finite number of functions. Further, in special cases, the interpolation functions have the discrete orthogonality. Since the measure of error is a convex function of the interpolation functions, it is ensured that the global optimum is obtained easily by using the ordinary numerical optimization. For some special cases, the concrete expression for the optimal interpolation functions are derived. Considering the approximation system in the reverse direction, where the linear circuits first passing the input wave are exchanged with the interpolation filters, it is shown that the new interpolation functions also minimize the same measure of error. As a direct consequence, a successive approximation of the interpolation functions is presented, which is suited to the applications such as the multiplex communication of the images of both directions.
A simple method for precise estimation of sinusoidal frequency
Authors: Noro Yuichi , Inomoto Kenji , Kuno Kazuhiro
Published: July 2007

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 are deduced from the DFTs of five samples. The frequency estimation error is also discussed with a computer‐simulated experiment which shows a 1 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.
An algorithm for embedding a biconnected planar graph to maximize the total sum of vertex‐ and edge‐weights on the exterior window
Authors: Kotani Ken , Masuda Sumio , Kashiwabara Toshinobu
Published: July 2007

Suppose that a biconnected planar graph G = (V, E) is given such that each vertex and edge are weighted. This paper proposes an algorithm for finding a planar embedding such that the total sum of weights of vertices and edges on the exterior window is maximized (among all possible planar embeddings of G. Among algorithm for determining planarity of a graph there is one that uses a data structure called a PQ tree. The algorithm presented in this paper is the one which extends it and uses the PQ tree as its data structure as well. However, it can find a required planar embedding in O(|V|) time by assigning special weights to each node of each PQ tree generated in the algorithm.
An algorithm for testing the planarity of a hierarchical graph
Authors: Kashiwabara Toshinobu , Masuda Sumio
Published: July 2007

Suppose that a graph G = (V, E) and a partition V1, V2, …, Vk (where Vi ∩ Vj = ϕ(i≠j), V1 ∪ V2 ∪ … ∪ Vk = V) of V into K subsets are given. If a vertex v belongs to a set Vi, the level of v is said to be i.

This paper proposes an algorithm which tests whether or not there is a planar embedding of G satisfying the following two conditions.

Condition 1. The vertices are arranged from top to bottom in the increasing order of their level (the vertices of the same level are placed on one horizontal line).

Condition 2. An edge crosses a horizontal line at most at one point. This algorithm uses a data structure called the PQ‐tree and it can be executed in O(|V|) time.
Fault tree analysis using disjoint disjunctive forms
Authors: Tajima Hisaaki , Iida Toshiyuki
Published: July 2007

In fault tree analysis (FTA) using minimal cut sets, the analysis time increases exponentially with fault tree size, thus analyzing a large fault tree is tedious. This paper proposes some algorithms to efficiently calculate the failure rate and structural importance of fault trees. Each fault tree is decomposed into subtrees, whose disjoint disjunctive forms are derived. The failure rate of the top event can be calculated with these disjoint disjunctive forms. Some examples confirm the efficiency of these algorithms.
The cooperative behavior of a compound system
Authors: Hirai Kazumasa , Shindou Hisanori
Published: July 2007

The behavior of a compound system made up of several subsystems may display a certain order due to the cooperation of each of the subsystems. The heart, made up of many different cells, is an example of such a system. When not connected to each other, the cells behave independently; but when connected, they cooperate to produce order in the total system.

The purpose of this paper is to investigate the transient behavior of each subsystem, and its contribution to the steady state in such a cooperative system. In order to investigate this, a multi‐inverted pendulum system will be studied, in which several inverted pendula attached to a small cart are controlled in such a way that they will stand inverted when the cart is moving. The results of this simulation produce an interesting motion which might not be logically deduced otherwise.
Algorithms for detecting separability of nonlinear mappings using computational graphs
Authors: Yamamura Kiyotaka , Kiyoi Masahiro
Published: July 2007

A nonlinear mapping is called separable (or partially separable) if it contains all (or some) variables in the separable form. Separability is a very valuable property of nonlinear mappings. Recently, many algorithms have been presented that exploit the separability of nonlinear mappings for improving the computational efficiency. It has been shown that the substantial improvement of computational efficiency can be achieved by exploiting the separability, especially when the number of separable variables is large.

To apply these algorithms, it is necessary to detect which variable is contained in the separable form. However, such algorithms have not been presented in the past. When the nonlinear mapping is large scale and complicated, it is difficult to detect the separability by inspection.

In this paper, some algorithms are proposed that detect the separability of nonlinear mappings automatically. For this purpose, the concept of computational graphs is introduced. Two algorithms are proposed that are based on the top‐down search and the bottom‐up search.

These algorithms are the combinations of simple techniques, but they can detect the separability in execution time proportional to the number of edges of the computational graph.
Evaluation of coded speech quality degraded by cell loss in ATM networks
Authors: Nagabuchi Hiromi , Kitawaki Nobuhiko
Published: July 2007

The asynchronous transfer mode (ATM) is considered interesting as the signal transfer mode in multimedia communication which can realize flexible and efficient channel usage. In ATM, the quality of speech is degraded by cellular handling of the signal which has a property different from the quality degradation in the conventional STM. Due to the small number of studies reported so far, the characteristics of the quality degradation have not been well understood.

This paper presents the design data for the speech quality in ATM network and considers the cell loss as the inherent factor degrading the quality in ATM. The complex effect of the cell loss and the coding distortion on the overall subjective quality and the articulation of the speech is measured and discussed. For several methods of speech coding and cell structures, the degradation of quality is examined when the cell loss rate increases.

It is shown that the 32‐kbit/s embedded ADPCM with the cell structure, where the lower two bits are selected with priority to suffer the loss, has a high immunity to the cell loss. Through the quality evaluation of the coded speech containing a cell loss, two factors are shown to be important: the speaker and the speech power at the site of cell loss. As a result of this study, new knowledge is obtained concerning the speech quality degradation characteristics in ATM network, and the data and knowledge useful in the speech quality design of ATM network are obtained.
An efficient three‐layer channel routing algorithm without vertical violation
Authors: Rhee Byungho , Sugai Yasuo , Hirata Hironori
Published: July 2007

This paper proposes a three‐layer channel routing algorithm for a new HVH routing method using the graph‐theoretic technique aiming at the minimization of the routing area. In the proposed method, the total net list is given as the input.

To realize the routing with the lower bound for the number of tracks, the following steps are executed. First, the net requiring the routing with priority is selected based on the channel graph. Then, using that net as the core, the net group that can be assigned to the present track is constructed using the graph‐theoretic technique. The proposed method has the advantage that no vertical violation is produced between different nets nor is the stacked contact produced essentially within the same net.

As a result of computer experiment for several well‐known problems, solutions with the minimum number of tracks are obtained. It should especially be noted that the vertical violation is not produced even in Deutsch's difficult problem. The routing can be realized within the lower bound for the number of tracks without the short vertical wire (where a short vertical wiring is employed in the horizontal wiring layer).
Generation of learning task sequences reflecting a fuzzy teaching strategy
Authors: Akahori Kanji
Published: July 2007

This paper proposes learning task sequencing in instructional design. The intention of the instructional designer in constructing the sequence is called teaching strategy. Previously, the author has proposed a teaching strategy model by introducing the concept of correlation [1, 2]. In this method, the subjective extent of the teaching strategy is represented by a number between 0 and 1. Such a numerical representation is somewhat difficult to determine from the viewpoint of the instructional designer. Consequently, a method is introduced which gives a subjective and fuzzy representation.

There can be more than one learning task sequence which reflects the teaching strategy represented by the subjective fuzziness, and a method is presented that derives all sequences. By this method, the learning task sequence can be determined under the subjective and fuzzy teaching strategy.
An efficient method for making a fault dictionary for short‐circuit fault diagnosis of linear networks
Authors: Sakaguchi Kazuhiro , Kaneko Mineo
Published: July 2007

In the fault diagnosis for the breakdown fault of the analog circuit such as short‐ or open‐circuit, the fault dictionary usually is employed which describes the input‐output characteristics for the accessible nodes for all assumed faults. In the construction of the fault dictionary, generally a large number of inverse matrix calculations is required, and the computational complexity becomes tremendous with the increase of the circuit scale. Thus, how to construct and maintain the fault dictionary efficiently is important.

This paper especially considers the short‐circuit fault in the fault diagnosis of the analog circuit and proposes a method which efficiently constructs the fault dictionary. This method considers the difference of the input‐output characteristics in normal and fault operations, for which the efficient calculation is presented. The method can be characterized as the extension of Johnson's proposal based on Householder's technique.

In the diagnosis by dictionary, the fault generally is diagnosed by comparing the items of the fault dictionary and the measured result. By the proposed method, the computational complexity in constructing an item of the dictionary can be reduced to the same order as that of the comparison procedure (except for the preliminary calculations in common to the whole item).
Influence of dimensions on the sensitivity characteristics of thickness extensional mode piezoelectric ultrasonic transducers
Authors: Kamata Hiroshi , Chubachi Noriyoshi
Published: July 2007

If the thickness‐to‐width ratio of the thickness extensional mode piezoelectric ultrasonic transducer is not too large, the mode coupling will produce a complicated resonance system, and this will complicate the sensitivity characteristics, which is a basic characteristic of the ultrasonic transducer. If the dimensional ratio is adapted in such a way that the lateral mode can be suppressed, so the single mode of the thickness extension can be simplified, then the sensitivity characteristics of the ultrasonic transducer can be simplified. On the basis of lumped parameter equivalent circuit of the three‐dimensional coupled piezoelectric transducer, the three‐dimensional coupling sensitivity vs. frequency characteristics are derived, and the relationship between dimensional ratio and receiving sensitivity is clearly provided. From the frequency condition of the maximum sensitivity, the antiresonance coupling factor of the parallel longitudinal mode and, thus, the dimensional ratio condition of the mode simplification are derived. Furthermore, the degree of coupling with receiving sensitivity of parallel longitudinal mode is used as a measure to define the sensitivity coupling factor; and, based on the sensitivity coupling factor, the condition to simplify the sensitivity characteristics of ultrasonic transducer is shown.
Design of FIR digital filters with specified tolerances—A neural network approach
Authors: Zhang Xi , Iwakura Hiroshi
Published: July 2007

This paper presents a design method for the equal‐ripple, linear‐phase FIR digital filter when the tolerance for the error of magnitude response in the passband or stopband is specified. To realize the design when various constraints are imposed on the transfer function or magnitude response, the design problem is formulated as the optimization problem.

A solution method is shown to solve the problem using the constrained neural net. When the lower bound of the magnitude error in the stopband is set as zero in the proposed method, the filter with double zeros in the stopband can easily be designed. Examples of application are shown for the minimum‐phase FIR filter and the matched filter pair with zero intersymbol interference.
Chaotic phenomena in an autonomous circuit with a nonlinear negative inductor
Authors: Nishio Yoshifumi , Mori Shinsaku , Inaba Naohiko
Published: July 2007

This paper proposed a new type of constrained circuit containing a complete saturation‐type nonlinear negative inductor and analyzes bifurcation phenomena and chaos generated in this circuit. The objective circuit is a three‐dimensional autonomous one and it is governed by the circuit equation which is similar to the constrained equation of the ideal diode type proposed by the present authors. Because this circuit equation is described piece‐wise by a two‐dimensional equation, a Poincaré map can be derived rigorously as a one‐dimensional map. By means of this map, the process of the generation of logistic map chaos can be explained in detail. Moreover, the analyzing results are verified by circuit experiments.
The identification problem and image reconstruction of the exponential X‐ray transform
Authors: Imiya Atsushi
Published: July 2007

The emission‐type CT which utilizes the radiation from the radioisotope (RI) injected into the body can reconstruct the biochemical information inside the body as an image. At present, the multicross‐sectional method is employed in reconstructing the three‐dimensional distribution of RI, as in the case of X‐ray CT.

In this method, a spatial nonuniform resolution is produced in the reconstructed RI image. It should be noted, on the other hand, that RI injected into the body emits radiations in all directions in the space. By utilizing all those radiations, a three‐dimensional image can be reconstructed without a nonuniformity in the resolution.

This paper shows that the three‐dimensional distribution of RI can directly be reconstructed from the radiations emitted spatially from inside the body. It is shown first that the problem to determine the three‐dimensional distribution of RI from the radiations is the inverse problem with the exponential line integral as the measured value. The relation between the Fourier transform of the projections by the exponential line integral and the Fourier transform of the original three‐dimensional image is analyzed. Then it is shown that the radiation attenuation coefficient of RI can be determined only from the projection data.

Furthermore, based on the relation between the Fourier and Hankel transformations, a method is derived for the direct three‐dimensional image reconstruction from the radiation projection using the series expansion.
A function generator based on quadratic B‐spline functions
Authors: Kamada Masaru , Toraichi Kazuo , Nakamura Norio
Published: July 2007

An improved smooth analog signal generator is proposed which purges digital‐to‐analog (DA) conversion phase distortions.

The author's previous system for such a generator had the following problems: 1) circuit parameter errors that accumulated at the circuit output due to the complicated circuit structure and 2) the output could turn out to be discontinuous because it was a subtle patchwork of discontinuous functions, even though theoretically they were smoothly connected. These problems have been avoided through careful tuning of the circuit parameters.

The present paper proposes an improved system, free of the above problems. This was accomplished through the following: 1) the analog circuit was simplified by retaining as many functions as possible in the digital circuit and 2) the analog output signal was made inherently smooth through a linear combination of time‐limited smooth (i.e., B‐spline) functions. Consequently, the present system produces more accurate linear phase characteristics with easier circuit parameter tuning. Its effectiveness was evaluated on a trial product whose performance was compared to that of the previous generator.
Realization of monolithic variable gyrators and their application to active filters
Authors: Ishibashi Yukio
Published: July 2007

With the development of IC and LSI technology, it is strongly desired that integration of the analog filter be realized. This paper proposes a method to construct the differential input voltage‐controlled current source (VCCS) and the differential input, differential output VCCS, using the conventional current multiplier type VCCS. The features of the proposed construction is that the VCCS characteristics can be varied by the control current and the transconductance is not affected by the absolute accuracy of the resistances as long as their relative accuracy is maintained.

A method also is presented in which the conventional LC filter is simulated using the gyrators composed of the proposed circuits. Then, by adding a resistor‐controlled current source circuit, a circuit construction also is presented whose control current can be determined by an external resistor so that the characteristics of more than one VCCS can be controlled.

To verify the effectiveness of the proposed method, the third‐order elliptic filter with a 0.25‐dB ripple is designed. As a result of computer simulation, it is verified that the characteristics change little by the ambient temperature of ‐10 to 60°C and that the cut‐off frequency changes little for ±12.5 percent variation of the supply voltage. Thus, it is shown that the analog filter can be integrated without using the tuning system which has often been employed.
A convolutional coded modulation scheme designed for controlling signaling probability
Authors: Saito Hidetoshi , Kohno Ryuji , Imai Hideki
Published: July 2007

Most coded modulation schemes, such as Ungerboeck's trellis coded modulation (TCM), are based on equiprobabilities of the signal space points. These schemes improve the bit error rate (BER) performance if all signals have the same norm. For signals with different norm, unequiprobable schemes result in much more improved of BER performance and larger coding gain than those of equiprobable ones. Consequently, a new convolutional coded modulation system utilizing symbol unequiprobability is proposed. Calderbank and Ozarow proposed another system using unequiprobable signaling [2]. In their system a shaping code is employed to save the average signal power without decrease of the coding gain. In the system proposed herein, to save the average signal power and to obtain more coding gain, a convolutional encoder designed in the real or complex number field is introduced to control the signal probabilities directly.

An optimum distribution of signaling probability is derived such that the minimum free Euclidean distance can be maximized. Then the convolutional encoder which can achieve the optimum distribution of signal points is derived theoretically. BER performance of the proposed system using such a convolutional encoder is evaluated by computer simulations.

Finally, it is shown that for the 16‐QAM system, the proposed scheme achieves about 1.0‐dB improvement in coding gain over Ungerboeck's TCM.
Oscillations in negative resistance oscillatory circuit containing periodically operating analog switch
Authors: Matsuki Michitaka , Mori Shinsaku
Published: July 2007

Some examples of the electronic circuit containing periodically operating switches are the switched‐capacitor circuit, which also are circuits switching the passive elements. This paper considers the nonlinear circuit containing a switch as an example of the circuit containing the periodically switched active element. The circuit containing a nonlinear negative resistor with N‐shaped voltage‐current characteristic and a resonant circuit is considered. In this type of circuit, even if an oscillation is not self‐excited, there arises an extraordinary phenomenon that an oscillation is produced for a certain range of the clock frequency when a switch is inserted between the nonlinear resistor and the resonant circuit and is operated periodically. This paper analyzes this phenomenon.

It is verified experimentally that there arises a hard oscillation in the low‐ and high‐frequency range of the clock. A theoretical analysis is attempted for this phenomenon. It is seen that even if a self‐oscillation is not produced in the circuit containing a negative resistance, there can arise an oscillation by periodically operating a switch. This is interesting since it may lead to the effective use of the negative resistance.
An error position recording method for optical disks
Authors: Suematsu Toshinari , Imai Hideki
Published: July 2007

In order to improve the reliability of optical disks, the nature of the data errors generated on optical disks must be studied in detail. To accomplish this, recording the positions of all errors generated on the optical disk, then performing various analyses based on them is an effective method. Because the optical disk has a large capacity and the error rate of the data is relatively high, the amount of error position data is tremendous and the error position data must be efficiently recorded. On the other hand, when the excess data format for highly efficient data becomes complex, the processing speed of the analysis diminishes. In this paper, we explain an error position data recording method that was applied to recording the error positions of optical disks. This method can efficiently record the error positions in a relatively simple data format. Then a mathematical model of error generation is used to examine the efficiency of this method when random errors and burst errors are generated. Furthermore, an example is presented where this method is applied to error positions measured on an optical disk and its efficiency is evident.
On the generalizability and adaptability of a self‐organization clustering
Authors: Miyanaga Yoshikazu , Tochinai Koji , Okumura Shinji
Published: July 2007

This paper constructs an adaptive clustering method based on the Mahalanobis distance and discusses its essential characteristics with respect to the adaptability and generalizability. From some experimental results, this method is shown to provide a sufficient adaptive clustering ability. The clustering method proposed in this paper is based on a self‐organization method by which every useful independent base can be obtained on a characteristic space. Thus, the user's effort of learning becomes minimum compared with supervised learning. In addition, clustering based on an adaptive method is performed to consider the time variation of the characteristics. Furthermore, to improve the generalizability, a set of clusters is reconstructed in every certain period and a system is constructed such that accurate cluster sets and recognition are obtained. Because this reconstruction is also processed in a self‐organizing way, it does not require learning from an external source. To evaluate the characteristics quantitatively, its effectiveness is shown by several simulation experiments.
Performance evaluation of a minimum bit error decoding algorithm of convolutional codes and its application
Authors: Ohashi Masayoshi , Yasuda Yutaka
Published: July 2007

Basic decoding characteristics of a minimum bit error decoding (MBED), originally proposed by Bahl et al., is investigated. It is shown that MBED can predict the bit error rate of each decoded bit with good precision. Then MBED is applied to the convolutional/Reed‐Solomon (RS) concatenated coding scheme. The outer RS codes are decoded by an erasure decoding using the symbol erasure information provided from the MBED results of inner convolutional codes. Simulation results show that the overall bit error rate is reduced one‐half to one‐third more than the conventional nonerasure decoding.