bluuebunny/crossref_metadata_2025 huggingface dataset:
Sampling theorems using nonuniform sample points and interpolation functions with band‐limited continuous spectra
Authors: Kida Takuro , Akagawa Keiichi
Published: July 2007

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, 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

This 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

The signal space composed of spline functions, which are smooth piecewise polynomials and almost free from oscillation, has attracted attention in the problems of interpolation and signal approximation. to make full use of such a signal space, there must be available a clear description of its properties. As a general approach to this problem, it is effective to derive the sampling basis in the signal space. Previously, the authors discussed the signal space composed of spline functions, and derived the sampling basis for the case where the degree is restricted to two. This paper extends the result to the arbitrary degree. 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.
Block placement method based on evaluation of idle area using simulated annealing
Authors: Koakutsu Seiichi , Sugai Yasuo , Hirata Hironori
Published: July 2007

The simulated annealing method is concerned with the large‐scale combinational optimization problem such as VLSI layout design. It aims at arriving at the global optimum without being captured by the local minimum, by the probabilistic optimization based on the stochastic process. This paper considers the idle area between blocks and proposes a method of handling the block placement problem, where the simulated annealing is applied to the optimization. The result of the proposed method is compared with the result obtained by the constant‐ pressure Monte Carlo method, which is one of the block placement methods with simulated annealing. It is seen that the proposed method can arrive at an almost equivalent result, even though the number of parameters is less and the determination of the parameters requires less time.
Transposition operations in permutation networks
Authors: Wu Dun , Kurihara Masahito , Kaji Ikuo
Published: July 2007

Numerous routing algorithms have been proposed for the overall network on the traditional Benes permutation networks. to achieve efficient system communication, a study of routing through a small number of information paths is needed from a practical viewpoint. In other words, it is desired that the requirement of permutation by a system should be realized so that for networks currently used, only a small part of the switch settings is affected. In this paper in the forementioned principle, a dynamic routing algorithm (basic design) based on permutation is proposed. Given a permutation t as a control input, a new permutation P̄ = P · t can be realized through currently realized permutation by resetting only a part of the information paths in a network, which does not affect any other information path. Moreover, as an element of a permutation network, we propose a d‐transposition block which performs a transposition operation efficiently, and show examples of d‐transposition blocks for d = 3, 4.
Two conjectures concerning nonlinear circuit problems
Authors: Itoh Makoto , Tomiyasu Ryuichi
Published: July 2007

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 fore‐mentioned 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.
Circuit module placement method based on physical model
Authors: Yamada Shoichiro , Fukunaga Kunio , Kasai Tamotsu
Published: July 2007

This paper proposes an efficient circuit module placement based on a force model. It is known that the module placement based on the force model arrives at a placement relatively close to the optimum. However, in the traditional method, a repulsive force is employed. As a result, the equation be comes nonlinear with a large number of variables, which is not easy to handle. First, this paper shows a simplification by replacing the forementioned problem by a nonlinear equation with a single variable. Then the module placement based on the foregoing formulation is proposed. Discussions are made on the size of the module, the case of multi‐terminal nets and the optimality of the obtained placement. Finally, a result of application to the placement design of the gate array is presented, indicating the use fulness of the proposed method. the features of the method are that the solution of the nonlinear equation can easily be determined, and the sparse matrix technique can be applied. Consequently, it is applicable to the placement design of a large‐scale circuit with more than 1000 modules.
An analysis of electrical equivalent circuit constants for a GT cut quartz crystal resonator with supporting portions at both ends
Authors: Kawashima Hirofumi , Matsuyama Masaru
Published: July 2007

A GT cut quartz crystal resonator with supporting portions at both ends has superior frequency‐temperature characteristics similar to the conventional rectangular GT plate. It was reported that it is a high‐Q resonator with a small equivalent series resistance from experiments [1, 2]. In this paper, the electrical equivalent circuit constants of the GT cut quartz crystal resonator with supporting portions at both ends is described. First, the equivalent circuit of the GT cut quartz crystal resonator is studied. It is found that the main vibration and the subvibration can be treated approximately as a single‐mode resonator since the coupling coefficient is only about 3 percent. Next, the relationship of the motional inductance L1, motional capacitance C1, parallel capacitance CO, capacitance ratio r1, and electromechanical coupling coefficient Kvl of the main vibration to the plate thickness electrode area and cut angle ϕ is found theoretically. Also found is the relationship of the motional inductance L2 of the subvibration and the plate thickness. These relationships are compared with the experiments. Calculated and measured results agreed well. Hence, the present analytical results are useful for design of a resonator with a new frequency.
Propagation characteristics of plate waves in a Z‐cut X‐propagation LiTaO3 thin plate
Authors: Toda Kohji , Mizutani Koichi
Published: July 2007

This paper describes the numerical analysis and experimental verification of the behavior of plate waves in a Z‐cut X‐propagation LiTaO3 thin plate. With the product of the frequency and the substrate thickness as a parameter, the phase velocity, mechanical displacement and electromechanical coupling coefficient are obtained. the Lamb waves of zeroth‐order antisymmetric mode (AO) and symmetric mode (SO) are coupled more tightly than the Rayleigh mode. the SO mode SH wave is a tightly coupled mode comparable to the piezoelectric plate, and is of low dispersion. the results of an experimental verification in an LiTaO3 of 100 μ thick agreed well with the theoretical data in the frequency range below 50 MHz. These results are effective in designing the plate wave devices.
Convergence properties of waveform relaxation‐newton method
Authors: Urahama Kiichi
Published: July 2007

This paper shows the following properties for the waveform relaxation‐Newton method: (1) the iterative solution converges uniformly and globally under the same condition as the convergence condition for the waveform relaxation method; (2) the waveform Newton method used in the inner loop of the iteration converges locally with the second‐order; (3) the waveform relaxation‐Newton method is slower in the convergence both globally and locally than the waveform relaxation method; (4) the time interval where the error monotone decreases is shorter in the waveform relaxation‐Newton method than in the waveform relaxation method; and (5) the convergence speed asymptotically approaches that of the waveform relaxation method.
Transient analysis of anisotropic elastic media by Bergeron's method
Authors: Sato Masahiro , Yoshida Norinobu , Fukai Ichiro
Published: July 2007

We have proposed a transient response analysis method for isotropic elastic media. In this method, the transient response analysis formulation by Bergeron's method is applied to the equivalent circuit of isotropic media made of multiple spatial combinations of local distributed equivalent circuits. This paper describes the basic treatment using equivalent circuit elements and their characteristics when the previous method is extended to the analysis of anisotropic elastic media. Since the equivalent circuits are used for analysis in this method, correlation with the electrical treatment of electron device design is excellent. By applying Bergeron's method, the transient response can be calculated successively at each discretization point similarly to the finite difference method. These characteristics are attractive from engineering viewpoints. the local distributed equivalent circuit representation of anisotropic elastic media and the formulation of the transient responses using Bergeron's method are also described. the results of the application of the present method to fundamental anisotropic problems are presented.
Energy trapping of higher‐harmonic width vibrations in piezoelectric strips and its application to filters
Authors: Watanabe Hiroshi , Shimizu Hiroshi
Published: July 2007

The energy trapping of the higher‐harmonic width‐extensional and width‐shear vibrations in a piezoelectric ceramic strip is studied based on the dispersion characteristics of these modes. By the use of energy trapping of the higher‐harmonic width vibrations, new small‐sized trapped‐energy resonators and filters that can be used at the high‐frequency range beyond several megahertz are developed. In the resonator and filter using a higher‐harmonic width vibration, it is necessary to excite strongly the width vibration of a specific order and to suppress all other vibrations. In this paper, a new electrode configuration is proposed for this purpose by taking into account the stress distribution of the mode. In addition, this method is used for fabrication of 4.5 and 10.7 MHz trapped‐energy filters using the 3rd‐harmonic width‐extensional vibration and the trapped‐energy resonators using the 2nd, 3rd‐ and 4th‐harmonic width‐extensional vibrations.
Super‐resolution NMR spectrum estimation method
Authors: Nishiyama Kiyoshi , Mita Tutomu
Published: July 2007

Several methods have been proposed to determine a high‐resolution NMR spectrum from limited data. This paper attempts to improve and extend the NMR spectrum estimation method using a complex Kalman filter, which was proposed previously by Nishiyama 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.
A synthesis of RC active filters simulating uniformly dissipative LC filters
Authors: Ishibashi Yukio , Nakahigashi Hideto
Published: July 2007

It is known that the active RC filter simulating the ladder LC filter has a lower sensitivity than the cascade active RC filter. A number of construction methods for filters of this king have recently been proposed. This paper considers the simulation type active RC filter. First, an LC filter is proposed as the prototype, which is obtained by adding losses to L and C in the conventional ladder LC filter. A method is described which realizes the relation between the node voltages and the loop currents of the circuit by an active RC filter. By the proposed method, the inverting integrator section in the traditional leapfrog structure can be realized by an active RC circuit with an operational amplifier, and the noninverting integrator section is realized by a voltage‐follower and a first‐order RC low‐pass circuit. It is shown that the number of operational amplifiers in the proposed circuit are approximately half the conventional leapfrog construction. Then the element sensitivity analysis is made, and the magnitude of the element sensitivity is discussed. Finally, a 5th‐order Chebyshev low‐pass filter is constructed and an experiment is made. the element sensitivity is calculated by a computer. It is shown that the element sensitivity in the proposed circuit is lower by 10 to 20 percent than the circuit with the conventional leapfrog structure.
Secret sharing scheme realizing general access structure
Authors: Ito Mitsuru , Saito Akira , Nishizeki Takao
Published: July 2007

As a method of sharing a secret, e.g., a secret key, Shamir's (k, n) threshold method is well known. However, Shamir's method has a problem in that general access structures cannot be realized. This paper shows that by providing the trustees with several information data concerning the distributed information of the (k, n) threshold method, any access structure can be realized. the update with the change of the secret trustees and the relation to the threshold graph are also discussed.
A parallel processing architecture for parameter estimation with recursive least squares
Authors: Ohkama Hiroshi , Miki Nobuhiro , Nagai Nobuo , Miyanaga Yoshikazu
Published: July 2007

The recursive least‐squares method (RLS) is often employed in the identification of the linear discrete‐time system, in which the parameters are time‐varying. This paper aims at producing the output which is an estimation of the system parameters, and derives a new algorithm which minimizes the RLS criterion with (time‐variable) forgetting factor. A parallel architecture is proposed which executes the algorithm. the derived algorithm is composed of two‐step processes, which are the Agee‐Turner PD factorization update theorem and the forward‐substitution for solving linear triangular systems. the proposed parallel architecture is composed of a large number of processing elements (PE) connected in a regular way. the communication is performed only between adjacent PE, and the architecture can easily be implemented by VLSI. It can be considered as a systolic array composed of PE with a queue (FIFO memory). By introducing PE with a queue, the two‐step processes can be executed in parallel for the first time. It is shown that by employing the proposed parallel architecture composed of (N2 + 5N − 2)/2 PE in the real‐time processing, the estimated parameters for any time can be produced as the output, in a processing time independent of the number of parameters of the estimation, even though the computational complexity is of the order of square of the number of estimated parameters.
Design of FIR partial response filters with equiripple stopband attenuation (class II, V)
Authors: Takebe Tsuyoshi , Matsumoto Toyoji , Chaisawadi Ake
Published: July 2007

This paper considers the design of transmitter and receiver FIR digital filters with partial response of class II and V for data transmission. the filters are required to band‐limit the transmitting and receiving signal, while maintaining the overall impulse response with zero intersymbol interference (ISI). Two cases are considered: 1) the case where the transmitter itself has the zero ISI impulse response; and 2) the case where the transmitter and receiver filters are matched. In the former, the length of the filter is set to be equal to the number of samples between two zero‐cross points, which are at a certain symmetrical distance from the center of the ideal impulse response. Then a linear‐phase transfer function is derived with an equiripple approximation to the ideal amplitude frequency response. In the approxiamtion, the weight in the stopband is set as less than 1, and ISI of less than 1. percent is obtained. In the design of the matched filter pair, the overall order is set in the same way as in the former. Applying the Herrman‐Schuessler design method for the minimum phase‐shift filter in the frequency domain, the transfer function is derived.

To realize ISI of less than 1 percent, it suffices to set the weight in the stopband less than 5 times in the passband. Although an effort was made to decrease ISI by successive approximation with the mirror zeros as the parameters, only a slight improvement was obtained. In either case, class V requires approximately a twice higher order than class II to realize the same attenuation and ISI.
Realizing an arbitrary conductance matrix via operational amplifier and its applications
Authors: Imai Yukio
Published: July 2007

This paper presents a realization of an arbitrary conductance matrix using operational amplifiers, together with some 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.
An algorithm for finding a forest in a planar graph‐case in which a net may have terminals on the two specified face boundaries
Authors: Suzuki Hitoshi , Akama Takehiro , Nishizeki Takao
Published: July 2007

River routing in VLSI and the one‐layer wire routing problem for printed circuit boards can be formulated as a problem of finding a forest in a planar (grid) graph. Given a planar graph G representing a wiring region and sets of terminals to be equipotential (that is, nets), we want to find vertex‐disjoint trees (i.e., a forest), each of which interconnects terminals of each net. This paper presents an algorithm for solving the forementioned problem in O (MIN {kn, n log n}) time when terminals of nets are placed on the two faces of a planar graph G, where n is the number of vertices of G, and k, that of nets.
System design of the figure operation hardware engine used for LSI layout design rule checking
Authors: Sakamoto Mamoru , Onodera Hidetoshi , Tamaru Keikichi , Nomura Toshihiro
Published: July 2007

One of the methods to perform the high‐performance LSI design rule check (DRC) on an engineering work station (EWS), is to employ the dedicated hardware engine. This paper considers the figure operation, which is the fundamental function of DRC, and describes the hardware‐oriented algorithm and the design of the hardware system for its implemen tation. the algorithm is obtained by improving the slit method and the vector method used in the usual software system so that they fit the hardware implementation. the hardware is composed of five functional blocks (VS, OP1‐OP‐3 and WL), and is implemented on four of 16‐bit microprocessors with 1 MByte main memory, together with FIFO memory. the blocks operate in parallel, realizing a high‐speed operation based on the three‐stage pipeline. It was verified by a software simulation that the hardware configuration is adequate.
Two‐dimensional filter banks containing prefilter
Authors: Ikehara Masaaki , Takahashi Shin‐Ichi
Published: July 2007

Two‐dimensional (2D) filter banks are used in subband coding of images, and are systems which split a signal into the narrow bands and/or resynthesize it. This paper presents a method of constructing 2D filter banks using 2D prefilter and 2D polyphase filter banks. First, the input signal is divided into four kinds of band on the 2D plane by 2D prefilter with IFIR (Interpolated FIR) transfer function. Then the 2D prefilter is designed so that the four kinds of band become 1. These resulting output signals are band‐split separately and synthesized by 2D DFT polyphase filter banks. Hence, its design method becomes very simple and both the required order and the number of multiplications can be reduced.
Sampling theorems for multidimensional waves with casual interpolation functions
Authors: Kida Takuro
Published: July 2007

This paper considers approximation formulas of the n‐dimensional wave f(X) expressed by the sum of sampled values of f(X) multiplied by certain n‐dimensional waves called interpolation functions. A systematic discussion is made to determine whether or not the approximate function converges to the original wave f(X). In this paper it is assumed that the interpolation functions satisfy the causality in regard to a certain axis in the n‐dimensional space Rn. Usually, that axis is considered to be the time axis. These discussions are important from the theoretical viewpoint, since it is closely related to the problem as to whether or not interpolation functions can be realized as impulse responses of physically realizable time‐invariant linear circuits. the set of sampling points is defined as a set of vertices obtained by sampling periodically all the vertices of the n‐dimensional parallel polyhedra periodically arranged in the space Rn. This definition of the set of sampling points includes most of the important arrangements of the sampling points, such as the hexagon and octagon lattices in the two‐dimensional plane. In this paper, it is assumed that the interpolation functions have supports, which are parallel translations of each other. the functional forms of the interpolation functions may be different. the set of f(X) considered in this paper is defined as the set of n‐dimensional waves whose Fourier spectra have the weighted power‐of‐p norm (p > 1) smaller than the specified positive value. It is shown first that, under the condition that the squared‐sum of the interpolation functions is bounded, the considered approximate expression does not in general converge to the original wave. It is shown that if the foregoing condition is relaxed, an approximate expression can be found with the impulse response of a causal time‐invariant linear network as the interpolation function, which converges to the original wave.
Conditions of convergence with respect to extended forms of interpolatory approximation for multidimensional waves
Authors: Kida Takuro
Published: July 2007

In this paper we consider a family of output waves obtained by impressing simultaneously an n‐dimensional input wave f(X) to the finite number of time‐invariant linear networks. A problem is considered, where the original wave f(X) is to be approximated using the sampled values of the forementioned output waves. A unified theory is presented with respect to the convergence of the approximate wave to the original wave f(X). the discussion given in this paper may be considered as a basis for the future theory of subband coding and multiplexing of the multidimensional signals, in the sense that we use the sampled values of the multidimensional waves transferred through the linear networks. the approximate wave h(X) for f(x) is a sum of the forementioned sample values of these output waves, being multiplied by certain n‐dimensional waves called interpolation functions. the set of sampling points considered in this paper includes most of the practically important arrangements of sampling points, such as hexagonal and octagonal lattices on the two‐dimensional plane. the set of f(X) in this paper is defined as the set of n‐dimensional waves such that the corresponding Fourier spectrum has the weighted power‐of‐p norm (p > 1) which is less than a given positive value. Under several assumptions, sufficient, necessary, and necessary and sufficient conditions are derived for the wave h(X) always to converge to f(X). These convergence conditions are presented which are imposed on the transfer functions of the time‐invariant linear networks through which f(X) propagates as well as the arrangement of the sampling points and the interpolation functions. They are closely related to the rank of the system of linear equations determined by those parameters.
Recognition of vowels and a semivowel using formant locus extracted by cubic spline
Authors: Matsumura Masafumi , Yamane Hiroaki , Fujii Katsuhiko
Published: July 2007

The purpose of this study is to recognize the semivowel /j/ and vowels from the continuous speech of an unspecified speaker. the format frequencies are used as the feature parameters, which correspond to the articulation by the tongue and the lower jaw. In the extraction of the formant locus, the continuity is emphasized. A pre‐processing is made which detects the missing formant or unnecessary peaks of the speech spectrum. Then the interpolation and smoothing are made using the spline function. A triangle on the formant plane F1‐F2 plane) is considered, where the vertices represent the articulations of /a/, /i/ and /u/. the triangle is transformed into an equiliateral triangle. By this transformation, the speaker differences of the positions of the vowels on the F1‐F2 plane are normalized. the point minimizing the rate of change of the normalized formant frequency is determined, and, based on the result, the segmentation and recognition are performed. the recognition rate of 92.8 percent was achieved for the vowels in the continuous speech uttered by ten speakers. Then based on the analysis of articulation by tongue and lower jaw in pronouncing the semivowel /j/, a recognition experiment was attempted by extracting the feature parameters for discrimination from the shape of the formant locus. As a result, the semi‐vowel recognition rate of 94 percent was achieved for continuous speech uttered by nine speakers.
A 1024 × 1024 channel‐switching lsi and its application to switching systems
Authors: Amada Eiichi , Kuwahara Hiroshi , Shirasu Hirotoshi , Shibukawa Masaru , Ikuzaki Kunihiko , Harakawa Takao
Published: July 2007

This paper presents a configuration of switching LSI based on 1.3 um CMOS technology, its function and application example to the channel switching. the switching LSI has a function to switch arbitrary time slots of the input and output highways. the switching capacity of 1024 × 1024 channels was realized by a single chip. the LSI includes the higher‐level functions of processor interface, but frame synchronization and fault monitoring. As an example of applications of this LSI, a switching network with 32768 channels (max) is realized.
Construction of variable cutoff digital filter based on a changeover of taps
Authors: Ikehara Masaaki , Takahashi Shin‐Ichi , Kato Mariko
Published: July 2007

This paper shows that by taking every M‐th impulse response from the impulse responses of FIR low‐pass digital filters and constructing a new digital filter based on them, a filter is obtained with low‐pass characteristic such that the passband width is M times that of the original filter. Next, taking advantage of this characteristic, a method is proposed for designing a low‐pass digital filter with variable cutoff frequency based on the same circuit by first designing a basic digital filter and varying M, that is, changing over taps. This principle is applicable not only to FIR filters but similarly also to IIR filters if some restrictions are allowed for denominator polynomials. Further, it is shown that by applying this theory to IFIR (Interpolated FIR) digital filters and multirate digital filters, filters can be realized which are more effective with respect to their cutoff characteristic and operational points.
A layout design methodology using data description with correspondence between circuit and layout
Authors: Okuda Ryosuke , Onodera Hidetoshi , Tamaru Keikichi
Published: July 2007

Problems at present in the manual layout design of logic LSI are the high desing 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.
High‐speed decoding of BCH codes using a new error‐evaluation algorithm
Authors: Horiguchi Toshio
Published: July 2007

A new error‐evaluation method for computing error values in decoding Reed‐Solomon codes or nonbinary BCH codes using the Berle‐kamp‐Massey algorithm is proposed.

Since the error‐evaluator polynomial used in the new method is obtained as by‐products of the computation for the error‐locator polynomial, the new method is more efficient than Forney's method whose error‐evaluator polynomial cannot be obtained as the by‐products of the error‐locator polynomial.
Analysis of sound fields in rooms using Bergeron's method
Authors: Shimoda Hidemaro , Yoshida Norinobu , Fukai Ichiro
Published: July 2007

At the designing stage of an auditorium, e.g., a concert hall or theater, it is imporant to predict its acoustic field via a computer. A conventional prediction method based on geometrical acoustic theory gives only a macroview of the field, and an addirional scale model test is required for a further investigation. to overcome this problem, although a technology to treat the sound field in a room as a wave field has been considered, there have been few examples of hree‐dimensional time‐response solution of the sound field in a room because of the difficulty of handling a wide analytical region, ide frequency band, and complex boundary conditions. This paper examines the justification of application of Bergeron's method to three‐dimensional sound field in a room. This paper also analyzes the correspondence between a three‐dimensional equivalent‐ciruit representation and the sound field, and examines the fundamental nature of a free sound field. Based on the results, a model of a simple rectangular room was considered, and its steady‐state sound pressure distribution (which is a fundamental characteristic the field) and simulation of reverberation (transient phenomenon) were examined. All the successful results show the usefulness of the proposed method.
An analysis method for acoustic waves composed of cosine waves
Authors: Tamaribuchi Kazuhiro , Saito Shuzo
Published: July 2007

This paper proposes a method of wave analysis which determines the frequency, initial phase and amplitude of the frequency components from three times the number of data compared with the number of components, using only the algebraic method. In the analysis, the frequencies are determined using the formula front sum to product of trigonometric functions, assuming the waveform is composed of a finite number of frequency components. Then using the determined frequencies, the initial phases and the amplitudes are determined. For verification, the signal of the in‐band multifrequency signaling system is considered as an example, and a computer simulation was made for the proposed frequency analysis. the signal is composed of one each from four kinds of low and high frequencies. Sixteen kinds of composite waves are constructed. the analysis is made as follows. Variations of ±1.3 percent are given to each of the low and high frequencies, and the initial phase was set as arbitrary in the range from ‐90 to +270 deg. in regard to the cosine wave. the amplitude is varied from 1 to 1/16. the sampling frequency is 4.48 kHz and the amplitude quantization is made in 12 bit. the maximum error in the result of analysis for the multiple frequency signal is ‐3.9 percent for 100, 000 trials, including the given frequency shift. In the analysis of wave composed of five components, the accuracy of nine digits is realized, using the double precision discrete data.
Equiripple magnitude approximation for low‐pass IIR digital filters with prescribed group delay
Authors: Nishikawa Kiyoshi , Choi Changseok
Published: July 2007

An efficient algorithm is proposed for approximating the magnitude of the low‐pass IIR digital filter with prescribed group delay in equiripple. Prescribed group delay characteristics can be realized by placing in the denominator of the transfer function a polynomial expression designed to have such characteristics. For amplitude characteristics we approximate it to obtain equiripple using a linear‐phase polynomial in the numerator.

Equiripple magnitude approximation is done separately for the passband and for the stopband and all the characteristics are made equiripple by several times of iterations.

In this simple method the auxiliary complex plane is utilized and in each the transformed transfer function is expressed as the sum of objective magnitude 1 and the error function for it. By assigning an all‐pass function to an approximation function of the error function, equiripple characteristics are realized.

By approximation in the stopband, good equiripple characteristics can be obtained even if all the degrees of freedom are used. On the other hand, since the approximation in the passband deviated largely from equirippleness, a method is presented for improving ripple characteristics by reducing the degree of freedom by one. As a design example, a filter of order 10 is given.
A multilevel data transmission baseband LSI using SC circuits
Authors: Nakayama Kenji , Ichihara Masaki
Published: July 2007

The baseband circuit for multilevel data transmission system was implemented by LSI using SC circuit technology. The waveform‐shaping filter, the full‐wave rectifier for clock regeneration, the narrowband filter and other circuits were constructed by SC circuits. The size and the power dissipation were reduced by constructing all filters in time‐division multiplex. The chip area and the power dissipation in the SC circuit is proportional to the total capacitance, and the total capacitance is determined by the capacitance ratio and the unit capacitance.

From such a viewpoint, the narrowband filter used in the clock regeneration is constructed by multirate, and the capacitance ratio was compressed using the N‐path filter. Capacitance partitioning is applied to high‐Q filters to suppress the capacitance ratio. Then by composing all capacitances as the integer multiples of the same unit capacitance, the desired ratio accuracy was achieved based on the unit capacitance of 0.1 pF. The problem of performance degradation due to integer capacitance ratio was minimized by the discrete approximation.

When the N‐path filter is composed by the time‐division multiplex scheme, the performance may be degraded greatly due to the parasitic capacitance among wirings. This effect was analyzed and an improvement from the viewpoint of the layout was proposed. As a result, the multilevel data transmission baseband LSI was realized with the chip size of 4.3 × 4/2 mm2 and the power consumption of 35 mW.
Numerical simulation of characteristics of condenser microphones at high ambient pressure
Authors: Suzuki Hisayoshi , Uchida Yoshiyuki , Oohashi Masanao , Nakai Takayoshi
Published: July 2007

The characteristics of a microphone are affected by the pressure and composition of the ambient gas. This paper presents a numerical simulation of the microphone within nitrogen gas with various pressures up to 48 atm. The simulation was carried out both for the deflection (static deflection) of the membrane by the polarization voltage and for the vibration (dynamic deflection) of the membrane by the incident acoustic pressure. In these cases, the physical constants of the gas, the location of the hole on the back plate and the gas flow between the back plate with a hole and the membrane were considered. The case also was considered in which the location of the hole is axially nonsymmetric. It was found that the membrane vibration derived by simulation corresponds well with the experimental result on the actual microphone as reported previously. From these simulations, the sensitivity frequency response of the condenser microphone under a high‐gas‐pressure environment were better understood.
Chaotic phenomena in an auto‐gain‐controlled oscillator using a pulsewidth controller
Authors: Uehara Tetsuo , Mori Shinsaku , Inaba Naohiko
Published: July 2007

This paper shows that in a certain type of amplitude‐controlled oscillator using a pulsewidth controller, correspondence of the Poincaré map with the cut map and logistic map can be obtained. It is proven almost analytically that the Poincaré map is ergodic if it becomes a cut map. By a computer‐assisted analysis, it is proven that chaos in the sense of Li‐Yorke is generated if it is a logistic map. Further, the validity of the analysis is confirmed by a circuit experiment.
Synthesis of LC simulation‐type RC active filters with reduced number of voltage followers
Authors: Ishibashi Yukio
Published: July 2007

It is known that the active RC filter which simulates the LC ladder filter has lower element‐sensitivity. However, recently, many such types of filters have been proposed. In this paper, a method of realization by low‐pass filter simulation‐type active RC circuit is described.

First, it is proposed that an LC filter to be simulated be realized by adding resistors in parallel to L's and in series to C' s of the conventional LC filter. Such a filter has the transmission zeros on the negative real axis of the 5‐plane, thus the conventional transfer function can no longer be used.

Next the derivation method of the transfer function for this filter will be presented. Then the realization of the LCR filter obtained by the active RC circuit using a reduced number of voltage followers will be described.

Finally, the synthesis and experiment of a fifth‐order Chebyshev low‐pass filter will be given. The effectiveness of the synthesis method is confirmed by the fact that the analysis of the element sensitivity by computer shows that in the passband it is below about 50 percent that of the leapfrog type circuit.
Maximum likelihood decoding of convolutional codes using channel states
Authors: Saito Osamu , Imai Hideki
Published: July 2007

Viterbi decoding and sequential decoding are well known as the maximum or approximate maximum likelihood decoding methods for the convolutional code. The decoding error rates of those decoding methods have been examined in the past. However, in those discussions the evaluation has been made by assuming the channel as memoryless. By contrast, this paper proposes several maximum likelihood decoding methods for the convolutional code in the channel model with memory, such as the Gilbert model. In decoding method (I), the conditional probability of the transmitted sequence for the received sequence is determined for each state of the channel. Using the result as the metric, the state of the channel is considered in the same way as the code trellis. The decoding is made by suitably selecting the metric. In decoding method (II), the state of the channel is estimated to suppress the computational complexity with the increase of the channel states. These methods are applied to Viterbi decoding. It is shown by computer simulation that the decoding error rate is improved over the traditional Viterbi decoding with equal complexity in instrumentation compared with the traditional decoding methods.
Fundamental studies on combined coding for information source and channel
Authors: Fukunaga Tetsuya , Murata Masashi , Kasahara Masao
Published: July 2007

From both theoretical and practical viewpoints, one of the most basic and important problems in the digital communication system is to minimize the distortion between the output signal from the information source and the destination signal on the receiving side, under the condition that the transmission rate in the channel is fixed. From such a viewpoint, this paper presents a detailed investigation of the trade‐off relation between the vector quantization (source coding) and the error‐correcting coding (channel coding), where the transmission rate of the channel is fixed as constant. The combined source and channel coding is shown, which is optimal in the sense that the average overall distortion is minimized in the communication system. The minimized average distortion is compared with the average overall distortion, where the ideal error‐correcting code with the minimum distance given by Hamming's upper bound is applied. A detailed discussion is made. The memoryless Gaussian source, the squared‐error distortion measure, the binary symmetric channel and the binary (n, k) block error‐correcting code are assumed. Finally, an investigation is made using the actual speech communication system.
CAM‐based hardware engine for geometrical problems in VLSI design
Authors: Suzuki Kei , Ohtsuki Tatsuo
Published: July 2007

A new hardware engine called CHARGE (CAM‐based Hardware Engine for Geometrical Problems) is presented to accelerate basic geometrical search algorithms for VLSI layout design and verification. Due to the world‐parallel nature of the Content Addressable Memory (CAM), basic geometrical search problems can be solved in constant time, independent of the number of geometrical data. This approach has flexibility to exclude complicated coding for various data structures depending on subproblems. The architecture of CHARGE and performance data measured on a prototype machine are described.
Design of three‐dimensional digital filters based on the outer product expansion of impulse response specifications
Authors: Ohki Makoto , Kawamata Masayuki , Higuchi Tatsuo
Published: July 2007

In dynamic image processing, medical image processing and seismic signal processing, high‐level processing, which cannot be executed by 1‐D or 2‐D signal processing, is expected to be realized by 3‐D digial signal processing. This paper proposed an efficient design method of 3‐D digital filters by the 3‐D outer product expansion which was proposed to decompose 3‐D arrays in image coding. In the proposed method, the given specification for the 3‐D filter is decomposed into the specifications for 1‐D filters using 3‐D outer product expansion and reducing the design of the 3‐D filter to that of the 1‐D filters. When the given impulse response specification is symmetrical, the computation required for the design in the proposed method can be reduced.
Analysis of parallel redundant system with dependent failures
Authors: Hidaka Toshimitsu
Published: July 2007

In communication line systems, in general, failure correlation (dependency of communicational line (path) composing a system due to the existence of common‐caused failure mode) often occurs and dependent failure correlation between communication lines (paths) must inevitably be considered in many cases. A concrete example is in reliability analysis between communication terminals in communication cable lines linking two central terminals, a characteristic quantity analysis of reliability when r‐fold common‐caused failures of neighboring lines occur. Thus, in this paper, we take a parallel redundant system given by common‐caused failure (r) and the number of communication lines (n), etc.