bluuebunny/crossref_metadata_2025 huggingface dataset:
A hardware implementation for a fast scaling algorithm in residue number system digital signal processing
Authors: Fujita Munenori , Kunieda Hiroaki
Published: July 2007

In residue number system digital signal processing, one major problem is the scaling operation. the general scheme for scaling has traditionally been to convert the number into an MRS (mixed radix system) format and then perform the scaling. This method includes problems such as the large number of hardware stages in the processing circuit and constraints on the selection of the scale number. This paper proposes a new method, where a direct modification of the Chinese remainder theorem is utilized to remedy the above problems. In the proposed scheme, the scale number can be set as a divisor of the module, and the residue number after scaling is determined by the operations in each residue number system. Using these changes, the memory capacity of the ROM is reduced and the ROM can be designed with 2 stages. Through these improvements the scaling speed of the circuit will also be improved.
Analysis of piezoelectric ceramic bending vibrators using interdigital electrodes both for poling treatment and for AC excitation
Authors: Oyama Sigeru , Hirose Seiji , Shimizu Hiroshi
Published: July 2007

The single‐plate ceramic bending vibrator, using interdigital electrodes for poling and excitation, has the remarkable features of high performance and reliability with little spurious response. This is in sharp contrast to the traditional bending vibrators with bimorph structure. This study aims at the description of the design principle for the bending vibrator with interdigital electrodes. A theoretical analysis is made using the conformal mapping, and a detailed discussion is presented on the relation of the capacitance ratio to the electrode dimensions and the poling voltage. Then the optimum condition is derived. A good agreement was observed by comparing the result of analysis and the experiment. It is shown that the capacitance ratio is less for the transversal coupling‐type bending vibrator, with the electrode finger being parallel to the direction of elongation, than for the longitudinal coupling type with the perpendicular direction.
LSI implementation scheme for general‐purpose high‐speed and coding rate viterbi decoder LSI
Authors: Kato Shuzo , Kubota Shuji , Ohtani Kouichi , Ishitani Tsunehachi , Miyahara Norio
Published: July 2007

This paper discusses the selection of the optimal device for LSI implementation of Viterbi decoder. The relation between the available number of gates and the performance of the decoder is described clearly. A method of reducing the number of gates for low‐and high‐speed LSI decoders, as well as their major performance, is described.

To reduce the cost for LSI as the system LSI, it is important to maintain the versatility of the developed LSI. From such a viewpoint, general‐purpose, a highly efficient Viterbi decoder is considered by using a ROM and two kinds of LSI. The construction and the performance are discussed, aiming at an LSI implementation.

The developed LSI operates at the frequency of 25 MHz or higher. It is applicable generally to the error correction with (N −1)/N coding rate. Compared with the Viterbi decoder using the same kind of LSI, the proposed LSI minimizes the number of chips and improves the speed by a factor of 1.5. By this elaboration, the device can drastically be reduced in size and cost, leading to an easier realization of a high‐speed, highly efficient and high‐gain error‐correcting system.
Analysis of a single server model with two queues having different service disciplines
Authors: Ozawa Toshihisa
Published: July 2007

This paper analyzes a single server model with two queues using the generating‐function method. Two service‐discipline models are used: a model in which the service discipline of one queue is “exhaustive” and that of the other queue is “gated”; and a model in which the service discipline of one queue is “exhaustive” and that of the other queue is “1‐limited.” For each model, the Laplace‐Stieltjes transform (LST) of waiting time distributions and mean waiting times are derived. By using the results, the difference of the mean waiting times in the two models is shown with numerical examples. It is possible to control the characteristics (e.g., the mean waiting times in each queue and their ratio, which depend on parameters such as the traffic density) of models with mixed service disciplines by changing the combination. Therefore, it is important to analyze these models for designing an integrated system which handles different traffic characteristics and required qualities.
Circuit simulation using waveform relaxation‐newton method
Authors: Urahama Kiichi
Published: July 2007

The following high‐speed technique is introduced into the waveform relaxation‐Newton (WRN) method: (1) the relaxation process starts from the predicted waveform; (2) the latent property in the time and the relaxation domain is utilized; and (3) iterative step width refinement, plus the window width adjustment to minimize the computation are proposed. A CMOS inverter chain was analyzed by the proposed variable window WRN method, and it was indicated that the window size control is useful not only in the Gauss‐Seidel‐WRN method, but also in the Gauss‐Jacobi‐WRN method, as well as in the case with a feedback. It is shown further that the computational complexity of the variable window WRN method is less than that of a similar waveform relaxation method and the interated timing analysis.
LSI and IC‐implementation of TDMA system—highly reliable and maintenance‐free TDMA equipment
Authors: Kato Shuzo , Morikura Masahiro , Umehira Masahiro , Enomoto Kiyoshi , Kubota Shuji
Published: July 2007

This paper aims for high reliability as well as reduction of hardware size, cost and development period for the time‐division multiplex access (TDMA) equipment, and proposes a new TDMA equipment architecture and modular construction. The fundamental approach employed in this paper is the digital realization of the analog circuit and the LSI implementation of the digital circuit. Then to realize TDMA equipment with a simple architecture and to realize a widely applicable LSI, TDMA equipment is reconstructed into a hardware‐oriented architecture. Based on the new architecture and the optimal function assignment to the modules, the following LSI and IC were developed. Six types for synchronization control, 11 types of LSI and IC for modulation‐demodulation (3 LSI, 4 MIC and 4 HIC), and 1 type of terrestrial network interface. By this development of LSI and IC, the hardware scale of TDMA equipment was reduced to one‐fifth, and the development period is reduced to one‐half compared with previous equipment. The maintenance‐free property is also achieved. The developed LSI and IC can be applied to various TDMA equipment in general with different TDMA frame and rate. For TDMA equipment to be developed in the future, the hardware‐scale and the development period can be reduced as in this case.
A method to construct constant weight t‐error correcting codes
Authors: Matsumoto Tsutomu , Imai Hideki , Saitoh Yuichi , Yamaguchi Kazuhiko
Published: July 2007

This paper proposes a construction method for constant weight t‐error‐correcting codes. These codes are nonsystematic t‐error‐correcting and all‐unidirectional‐error‐detecting (t‐EC/AUED) codes. The format of these codes is given by generalization of the Bose‐Rao code and the Kundu‐Reddy code, and a codeword of these codes is the concatenation of constant weight codes. Concrete methods to construct 3‐ and 4‐EC/AUED codes also are given. For example, 3‐EC/AUED codes of length 80 proposed so far have at most 246 codewords, whereas the proposed method gives a code having 251 codewords when Bose's method for encoding binary information words into constant weight words is used.
An efficient integer labeling method for solving systems of nonlinear equations with separable mappings
Authors: Yamamura Kiyotaka , Ochiai Makoto
Published: July 2007

The integer labeling method is one of the simplicial‐type homotopy methods for solving systems of nonlinear equations. In this method, matrix operations are not necessary and an approximate solution can be generated only by function evaluations. Thus, the algorithm becomes quite simple and can easily be programmed.

However, since the integer labeling method uses simplicial subdivision, the amount of computations grows exponentially with the system size.

For the vector labeling method, which is another type of homotopy methods, an efficient algorithm using a rectangular subdivision has been proposed.

For the integer labeling method, however, such an efficient technique has not been proposed because integer labeling on rectangles is essentially impossible.

This paper proposes an efficient integer labeling method for solving systems of nonlinear equations with separable mappings. Similarly to the conventional method, the proposed method labels integers on simplices. However, by making use of the separability and the concept of the rectangular subdivision, the number of function evaluations is reduced significantly.

In the proposed method, function evaluations are not necessary as long as the labeled simplex is moving in the identical rectangle, and when it moves into another rectangle, only one component of the function in that direction needs to be computed. Hence, with the increase of the system size, considerable improvements of the computational efficiency can be achieved.
The structure of the group of transforming rules on rotation algorithm of a dot matrix
Authors: Nakamori Mario , Hagiwara Yoichi
Published: July 2007

Assume that an n x n dot matrix is stored in consecutive n words of a computer with one word being n bit. This paper discusses eight operations to rotate or transpose the dot matrix in horizontal/vertical/slant direction. As is well known, such operations form a group.

This paper presents a unified description of the algorithms, and it is shown that all eight algorithms can be derived by suitably setting the constants in the unified algorithm. Then it is shown that the algorithms can be transformed to each other by simple transformation rules for the constants, and those transformation rules are shown to form a group.

The foregoing algorithms for the dot matrix are not unique, and there can be different descriptions for the algorithm producing the same result. Consequently, there can be various versions of the eight algorithms as well as versions of transformation rules. This paper shows that one of the two groups formed by the transformation rules is isomorphic to the group of algorithms, but the other is not isomorphic.
A dimensional analysis on chaotic neural networks
Authors: Ikeguchi Tohru , Itoh Susmu , Utsunomiya Toshio , Aihara Kazuyuki
Published: July 2007

We apply a dimensional analysis to a neural network model with chaotic dynamics, or a mathematical model of chaotic neural networks. First, Lyapunov spectra are examined to confirm existence of chaotic network behavior and to calculate the Lyapunov dimension. Then, the correlation dimension is estimated in the following three cases: (1) reconstruction from single variable time series, (2) observation in state subspaces and (3) observation of the macroscopic average activity. The results show good correlation dimension obtained by observation in state subspaces and the characteristics of the correlation dimension estimated by macroscopic observation.
High resolution spectral estimation using complex coefficient filter banks
Authors: Shimamura Tetsuya , Takahashi Shin‐Ichi
Published: July 2007

This paper considers first, for the purpose of estimating the frequency of sinusoids corrupted by noise, a spectral estimation method employing the analytic signal, and proposes an analytical signal pre‐filtering technique for that estimation method. Also, it is shown that the analytic signal processing in spectral estimation gives rise to reduction in the estimation variance.

Next, the spectral estimation method with complex‐coefficient bandpass filters is considered and a method with complex‐coefficient filter bank is developed which could analyze the entire frequency region which contains any simusoids to be estimated. This method with filter bank could resolve a problem found in the conventional method with decimation techniques. Furthermore, an algorithm is obtained which could provide considerably more accurate estimation results than the conventional one. This algorithm effectively employs the signals remaining in the sampling operation of decimation, and then averages them.
A computational algorithm for solving 0‐1 goal programming with GUB structures and its application for optimization problems in system reliability
Authors: Gen Mitsuo , Ida Kenichi , Lee Jae‐Uk
Published: July 2007

Upon to now, system optimal allocation problems such as system reliability and system availability problems have been formulated as single‐objective problems and solved through the use of various well‐developed optimization techniques.

However, in this field, there are many problems that cannot be solved without applying MODM (multiple‐objective decision making) methods. These methods deal with multiple objectives that conflict with each other instead of formulating the problem as a single objective programming problem which optimizes only the reliability of the cost function, as is done in previous methods.

GP (goal programming) is one of the most powerful MODM tools in this field. In practical MODM problems, many GP problems involve a large number of 0‐1 decision variables and a special type of 0‐1 variable, which arises during the transformation of non‐linear integer programming into 0‐1 linear programming.

In this paper, we propose an efficient and specific algorithm for solving large‐scale 0‐1 GP problems in particular structures, which are termed GUB structures. Furthermore, to illustrate the effectiveness of the algorithm proposed here, we introduce two numerical examples from among the problems of system reliability, and compare the algorithm proposed with previous methods.
An optimal backup policy considering jobs for a hard disk
Authors: Sandoh Hiroaki , Kawai Hajime
Published: July 2007

The advent of the present‐day low‐cost personal computer and work station has seen widespread applications. These personal computers and work stations often are used as a system including the hard computer disk (called the hard disk). Various files are constructed and stored in the hard disks. Often, those files are lost due to the mis‐operation of the user or the failure of the hard disk itself.

The usual means to cope with such a situation is to back up the content of the hard disk at specified intervals by the floppy disk and other devices. However, time to backup usually is left to the intuition and the experience of each user. This paper considers this situation, and proposes a backup policy where the backup is executed when a certain number of jobs are completed. Then a formulation is presented to determine the optimal number of jobs, in the sense that the expected cost per unit time is minimized.

In this paper, it is assumed that only those parts which are modified or updated after the previous backup are backed up at each time. Based on the formulated expression, the existence condition for the optimal solution is derived.
Certain types of Runge‐Kutta‐type formulas and reproduction of orbits of linear systems
Authors: Maeda Shigeru
Published: July 2007

Sanz‐Serna derived a condition for a Runge‐Kutta method to give a symplectic mapping when applied to an arbitrary Hamiltonian system. The methods with the condition have a very interesting property even if the system to be integrated is not Hamiltonian.

This paper discusses numerical reproduction of the phase portrait of a linear systemwhen using a Runge‐Kutta method which satisfies one condition in addition to that of Sanz‐Serna. Not only is such a method Astable, but its absolute stability region also coincides completely with the left half plane of a complex plane, and, accordingly, either convergence or divergence of atrue solution is reproduced numerically whatever the value of astepsize.

Furthermore, it is shown that every elliptic closed orbit is drawn accurately by the numerical method as long as the calculation error is ignored. An explicit Runge‐Kutta method, which usually is employed, does not have these properties. Finally, a four‐dimensional linear system is designated which admits all elliptic orbits, and convergent and divergent solutions. It is illustrated that its phase portrait is reproduced well by using the implicit midpoint method—one of the methods with the forementioned conditions.
A note on error estimation for spline interpolation method with sampling bases
Authors: Kamada Masaru , Toraichi Kazuo , Ikebe Yasuhiko
Published: July 2007

The spline interpolation using the sampling bases has a feature of real‐time computation which is not observed in other methods since the result of interpolation is derived directly from the sampled value sequence. Then it is necessary that the sampling function should be truncated in a finite interval. This paper aims to clarify the relation in this method between the error in the result of interpolation due to the truncation and the length of the truncation interval. The result is summarizes in a correspondence table for the truncation interval length and the error. In the proposed interpolation method, the truncation interval length to ensure the given tolerable range of error can be determined using the correspondence table.
LSI module placement methods using neural computation networks
Authors: Date Hiroshi , Hayashi Terumine
Published: July 2007

When solving combinatorial optimization problems using the neural networks proposed by Hopfield, the objective function and constraints of (0, 1)‐variable optimization problems must be transformed. Then the problem is that there is no theoretical formula to decide the weighting factors of the terms of the objective function.

This paper presents an LSI module placement algorithm which uses neural networks. Then an efficient method is proposed to decide fully adaptive parameters, where these parameters ensure convergence of the solution. Experimental results show that the new algorithm gives better solutions than the min‐cut algorithm, although it is not as good as the simulated annealing algorithm.
Design of a pipelined plane‐rotation algorithm based on the error analysis
Authors: Kikuchi Hisakazu , Nakashizuka Makoto , Watanabe Hiromichi
Published: July 2007

This paper aims at the realization of a discrete‐time lossless scattering system, and proposes a plane‐rotation algorithm in which the pipelined processing can be applied. In the proposed method, the CORDIC algorithm is divided into unit operations. The operational cells realizing the individual unit operation are connected in cascade to reconstruct the CORDIC algorithm into a pipeline. This configuration by the cascade connection realizes the pipelined processing and has a high modularity since the structure is a linear array of the unit cells.

Next, the operational error in the proposed algorithm is classified into the error due to the finite operation and the error due to the finite word‐length. A design for the algorithm is presented based on the result of analysis. The feature of the proposed design method is that the necessary operations can be extracted from the specified range of the finite operation error and the rotation angle, while minimizing the finite word‐length error of the algorithm. Finally, application examples are shown for the wave digital filter and the sinusoidal pair oscillator.
Log‐depth circuits for elementary functions using residue number system
Authors: Okabe Yasuo , Takagi Naofumi , Yajima Shuzo
Published: July 2007

A new hardware algorithm for evaluating elementary functions internally using residue arithmetic is proposed. The algorithm is an improved generalization of Beame‐Cook‐Hoover's log‐depth divider for evaluation of all elementary functions.

The proposed circuit computesn‐bit elementary functions in time proportional to logn. On a residue number system, a polynomial can be evaluated at each residue digit independently. Utilizing this property, each residue digit of an approximation polynomial of the target function is looked up from a small table in parallel. Thus the circuit is faster and smaller than Beame‐Cook‐Hoover's although both have the same order of computation timeO(logn). The circuit based on the algorithm is composed of three parts, i.e., binary‐to‐residue converter, ROM for table looking‐up and residue‐to‐binary converter. It has a regular structure suitable for VLSI implementation. By changing only the contents of looking‐up tables corresponding to the approximation formula, a circuit can compute any elementary function uniformly.
Realization of 2‐D separable‐denominator digital filters for real‐time and parallel processing
Authors: Hinamoto Takao , Muneyasu Mitsuji
Published: July 2007

This paper considers the 2‐D separable‐denominator digital filter, and discusses its real‐time and parallel‐processing‐oriented realization. The aim of this paper is to employ a large number of processors concurrently, and to realize a real‐time processing minimizing the data throughput delay by applying pipeline processing. More precisely, the case where the poles of the 2‐D separable‐denominator transfer function are composed of real and complex poles is considered, and two new methods of realization are presented based on the Roesser‐type local state‐space model, which is suited to the real‐time processing.

It is shown that the data throughput delay in those methods is equal to the time for two multiplications and one addition, or the time for one multiplication and two additions. The processor factor and the efficiency in each realization are investigated, and the effectiveness of the realization methods are verified from the viewpoints of real‐time processing and parallel processing.
An orthogonal function‐type adaptive digital filter applying a modified discrete fourier transform
Authors: Itoh Yoshio , Kobayashi Masaki , Morkawa Toshiyuki , Kusaka Hiroji
Published: July 2007

An orthogonal function‐type adaptive digital filter (orthogonal function type ADF) is known as a means of improving the convergence speed of a tap coefficient in the case in which input is a colored signal such as a voice. Two different methods have been used for pole location of a transfer function of this ADF. One is the pole location of equal interval of 2π/N(rad). (N: tap number of ADF) starting at the phase 0 (rad) on the circle whose radius is very close to 1 within the unit circle of the z‐plane. The other is the one resulting from rotating the above location by π/N(rad). In the case of the former pole location, it has been shown that it is equivalent to a frequency sampling type ADF such that input is a periodic process of period N. Why improvement in convergence speed of the latter is the case is not clear at this date.

In this paper we first derive an equivalent circuit of an orthogonal function type ADF using the latter pole location. In this circuit a modified discrete Fourier transform (MDFT) appears in which phase of a pole of a resonator of frequency sampling filter is shifted by π/N(rad). Next, we present a condition for expansion coefficients of the MDFT circuit to be orthogonal to one another. Moreover, we clarify that this equivalent circuit is of a construction that satisfies the orthogonal condition of the MDFT circuit. Finally, we verify these results of investigation by several computer simulations.
Multilevel coding for impulsive noise
Authors: Ogiwara Haruo , Kaneda Kazuyuki
Published: July 2007

Based on the result of measurement of the balanced pair, the model for the amplitude distribution of the impulsive noise is derived and the optimization of the multilevel coding for the impulsive noise is discussed. It is shown first that in the multilevel coding proposed previously, the burst error is produced at the input of the error‐correcting decoder, even if the noise is random. An improvement by interleaving is proposed.

Next, a theoretical approximate expression is derived for the error rate at the decoder output. This theoretical expression can be applied not only to the impulsive noise, but also to the noise with any amplitude distribution.

Finally, using that expression, the error rate for various assignments of the error‐correcting capability to each level is calculated, and the assignment realizing the maximum coding gain is determined. It is shown that the coding gain of 22 dB is realized when the baseband 4‐level transmission is encoded into 8‐level transmission, and the BCH code of code length 31 is employed.
A study on convergence speed of an orthogonal function‐type adaptive digital filter
Authors: Itoh Yoshio , Kobayashi Masaki , Morikawa Toshiyuki , Hayahara Eturo
Published: July 2007

An adaptive digital filter (orthogonal function‐type ADF) composed of an orthogonal function system which can locate poles of a transfer function in arbitrary positions in the unit circle in the z plane is known as a way to improve convergence speed in the case in which input is a colored signal (such as voice). However, no sufficient theoretical investigation has been done on the reason why this ADF rapidly converges to the colored signal input.

This paper derives an equivalent circuit of an orthogonal function‐type ADF in the case in which poles of a transfer function are located in equal space on the unit circle in the z plane. Then the convergence speed using this equivalent circuit is considered. It is shown that the orthogonal function‐type ADF is equivalent to a construction due to a periodic process of an input of the known ADF using the frequency sampling filter (frequency sampling‐type ADF). In other words, even if input is colored like voice and aperiodic process, in the orthogonal function type it is shown that orthogonality between tap coefficient inputs is preserved. Furthermore, these studied results are verified by computer simulations.
Design of subband transmission systems in time and frequency domain
Authors: Kida Takuro , Yoshioka Leopoldo Rideki
Published: July 2007

Consider a system in which N time‐invariant linear circuits Hk(ω)(k=1∼N) are connected in parallel and to each of which M parallel time‐invariant linear circuits Hkm(ω)(k=1 ∼N; m=1∼M) are connected in cascade. Then consider that a certain class of band‐limited signals f(t) is impressed on such a system, and the equal‐interval sampled values gkm(nT+akm)(k=1∼N : m=1 ∼M : n=0, ±1, ±2,…; T>0; akm of the output signals gkm(t)(k=1∼N; m=1∼M) are obtained. By multiplying the sampled values by the interpolation function Økm(t)(k=1∼N; m=1 ∼M), which is obtained by the convolution of the time functions Økm(t)(k=1∼N; m=1 ∼M), and by summing up, an approximation formula y(t) is determined.

A unified investigation is made on the approximation problem using y(t) for the response g(t) when the original signal f(t) passes through a given filter H(ω).

First, the error e(t)=g(t)‐y(t) is considered for given Hk(ω)(k=1∼N), and Hkm(ω)(k=1∼N; m=1∼M) over the set of f(t) belonging to the considered set of waveforms, and a formula is derived which represents the upper bound ∼max(t) of the error in a form where the interpolation functions Økm(t) and Øk(t) are separated.

Then, based on the derived formula, an effective design method for the interpolation filter is established, based on the coefficient equation and linear programming, assuming that all of the interpolation filters are constructed as FIR type.

Finally, the optimization of the system with a hierarchical structure is discussed.
Analysis of vehicle behavior at a merging section of APM system with different strategies
Authors: Hamamatsu Yoshio , Yoneyama Tadaaki , Kikuchi Shinya
Published: July 2007

This paper considers the merging section of an APM (automated people mover), which is an approach to the new traffic system. The behavior of the vehicles at the merging section is modeled, and the situation, when different strategies are adopted, is analyzed using the technique of Markov chain. An APM system is considered, where the vehicles in the system do not always run individually, but sometimes run in the form of a train.

This paper attempts to point out the essential dependencies of the vehicle behavior on the control strategy. It is assumed that the vehicles arrive individually. The basic idea of the control strategy is that the individually incoming vehicles are connected at the merging section. Thus the vehicle capacity of the outgoing line from the merging section is equivalently increased.

As the result of analysis, the relations among the arriving traffic of the vehicles in two incoming lines, the average waiting time of the vehicles in the individual lines, the average number of connected vehicles in a train, and the density of train operations are derived explicitly. The dependencies of the average waiting time of the vehicle and the outflow traffic density of the train on different control strategies are examined.
G‐type quartz crystal resonator with supporting portions in length or width direction of vibrational portion
Authors: Kawashima Hirofumi , Matsuyama Masaru
Published: July 2007

The objective of this paper is to provide a miniature quartz crystal resonator with a frequency around 1 MHz. Here, a new G‐type crystal resonator made of the vibrational portion and the supporting portion is presented. The frequency constant, frequency‐temperature characteristic and electrical characteristics are investigated theoretically and experimentally. The study is conducted with a view to realizing a small quartz crystal resonator superior in the frequency‐temperature characteristics and the electrical characteristics.

First, the vibration analysis or the experimental investigation is carried out for the case of supporting portions in the length or width direction of the vibrational portion. It is understood that there exists a zero temperature coefficient in either case and that the series resistanceR1is smaller for the structure with a supporting portion in the width direction. With the cut angles ø =‐60 ∼−70°, and θ = 45° in the present resonator, the frequency constant is much smaller than the one in the GT cut. It is shown that a small quartz crystal resonator can be realized at frequencies near 1 MHz.
A method to minimize sensitivity in cascade SC filters
Authors: Ma Zhiqiang , Nakayama Kenji , Yamamoto Gaishi
Published: July 2007

When a switched capacitor (SC) filter is made in an LSI form, it is important to reduce the total capacitance for reduction of the chip size, improve high‐frequency operation, and reduce power consumption. The total capacitance can be reduced by reducing the total capacitance when the minimum capacitance, is normalized to unity and y is the miniaturization of the capacitors on the chip. When the capacitor is made smaller, the error in the capacitance ratio increases so that the transfer characteristics of the SC filter are varied. To minimize the effect of this error, a circuit with a low element sensitivity is needed.

In this paper, for a cascaded SC filter frequently used as a standard circuit, general conditions have been derived that must be satisfied by the pairing and order of the zeros and poles. First, the element sensitivity of the transfer function of the second‐order SC circuit is derived. Next, based on this result, the formula is derived which represents the element sensitivity of the transfer function of the cascaded SC filter. Based on this formula, the condition is derived for the pairing and order of the zeros and poles to reduce the element sensitivity.

From the present study, it is found that the element sensitivity can be reduced by pairing nearby zeros with a high Q pole and that the element sensitivity is independent on the order of poles and zeros. It is found also that the element sensitivity increases in proportion to the cutoff frequency of the filter. The foregoing results are confirmed by the simulation using 11‐th order low‐pass and high‐pass filters.
Design of FIR fan filters using FIR hilbert transformer
Authors: Nishikawa Kiyoshi , Yamamoto Eiko
Published: July 2007

A new, more direct method for synthesizing FIR fan filters and quadrantal fan filters using a FIR Hilbert transformer as the one‐dimensional (1‐D) prototype filter is described. Only the case 3 transfer function of the Hilbert transformer is available. When an equiripple design is used for the transformer, the ripple amplitudes in the passbands and stopbands of the resulting fan filter are nearly the same as the ripple amplitude of the transformer. The transfer function of the fan filter has only a few nonzero coefficients because the Hilbert transformer is required to be of even order. Illustrative examples of fan filter synthesis are presented.
Design of layered‐neural nonlinear filters using backpropagation algorithm
Authors: Arakawa Kaoru , Harashima Hiroshi
Published: July 2007

The signal such as an image signal is a non‐Gaussian signal with an abrupt change. When a random noise is superposed on such a signal, it is impossible to eliminate the noise effectively by using the traditional linear filter. Then it is desired to develop a nonlinear filter.

From such a viewpoint, this paper proposes a nonlinear digital filter using a layered neural network, aiming at the elimination of such a noise. By thus introducing the nonlinearity and the learning ability of the layered neural network, the noise can effectively be eliminated. In this paper, the noise elimination is shown for the case of a one‐dimensional test signal and a two‐dimensional image signal indicating the usefulness.
A modification of the generalized minkowski metrics
Authors: Yaguchi Hiroyuki , Ichino Manabu
Published: July 2007

The authors have reported previously on the generalized Minkowski metrics which are applicable to the data with the mixed quantitative and qualitative features [1]. The metrics can simultaneously handle the ordinary quantitative features as well as the qualitative features with a finite set as the value. It is possible also to handle directly the quantitative features with the values being given by intervals. On the other hand, it is seen that a value sometimes appears which does not fit the human intuition. This paper defines the generalized Minkowski metrics, which have a natural property also for the data with the interval as the value. The usefulness of the approach is demonstrated through examples.
Operational state analysis using piecewise‐linear diode model with a switch and its evaluation
Authors: Tanaka Nobuyuki , Hiyoshi Yasunori , Suzuki Seiji , Asai Hideki
Published: July 2007

One of the methods for analysis of the operational state of the transistor network is the method based on the piecewise‐linear diode model. When the bipolar transistor network is analyzed using the Ebers‐Moll model containing the diode model composed of the voltage source and the switch, it may happen that a cut‐set composed only of current sources or a loop composed only of voltage sources is formed in the course of iteration. Also, depending on the state of the switch, it may happen that a subnetwork is separated. In such cases, the coefficient matrix of the network equation becomes singular.

From such a viewpoint, this paper discusses the method in which the modified nodal equation is constructed using the piecewise‐linear diode model containing a switch, and a high‐speed analysis is executed while avoiding the singularity problem by simultaneously changing the states of more than one switch. In the piecewise‐linear diode model, the reverse‐biased characteristic is approximated by connecting a linear resistor R in parallel to the switch branch so that the electrical characteristic of the diode is not violated. The forward‐biased characteristic is approximated by inserting a linear resistor r in series to the switch branch. It is verified that by using the piecewise‐linear diode model containing a switch, the singularity problem is avoided and the limit cycle problem is greatly ameliorated.
A grid‐free maze router
Authors: Kimura Mitsuyuki , Shiraishi Yoichi , Himeno Fumio , Hanaki Soichi , Kusaoke Manabu
Published: July 2007

Several maze methods using analog LSI as wiring path searching of grid‐free wiring have been reported. These techniques process in microns and the searching range is limited to prevent increases in processing time and used memory capacity. This paper proposes a grid‐free wiring technique for determining the shortest wiring path limits processing time and used memory capacity to within a practical range. This technique takes the conventional wiring grid interval, determined so that the through‐hole used to connect the wiring layers will not violate the design rules and divides the grid into l/n (n: arbitrary integer) and also controls the integer n. It allows for independent layer direction, variable width wiring, and rectangular terminals. A wiring technique is proposed that obeys wiring constraints, such as T‐shape wiring which arise from analog circuit characteristics. This technique was applied to four kinds of modules of analog LSI and improvements in the wirability and its effectiveness were confirmed.
SC pitch extraction for real‐time and continuous processing
Authors: Yamashita Shinsuke , Ogihara Akio , Yoneda Shojiro
Published: July 2007

If speech recognition is applied to man‐machine interface, real‐time processing is naturally desired. It is efficient for real‐time processing that routine preprocesses (extraction of features and analysis) are executed with custom hardware.

This paper proposes two kinds of pitch extractors which continuously extract the fundamental period of an input signal in real time. Since the pitch extractors are constructed using the switched capacitor (SC) technique, they can be integrated into an LSI package.

The first SC pitch extractor is based on the autocorrelation function, and the extractor calculates two variable multiplication with a programmable capacitor array. The second SC pitch extractor is based on the absolute difference function which is obtained by replacing the multiplication in the autocorrelation function with subtraction, and the absolute difference function is useful to reduce circuit scale.

A remarkable feature of both SC pitch extractors is that the pitch period of an input signal is continuously extracted in real time. Hence the pitch synchronous analyzer which performs spectral analysis in real time can be constructed with the proposed SC pitch extractors and the SC DFT circuit. According to the experimental results, the error of the SC pitch extractors made with discrete parts is less than 3.0 percent. In addition, the proposed SC pitch extractors were applied to five Japanese vowels and obtained a satisfactory result.
Abnormal convergence of numerical solutions and Runge‐Kutta‐type methods admitting no such phonemenon
Authors: Maeda Shigeru
Published: July 2007

When the standard Runge‐Kutta method is applied to a certain system of linear differential equations, the numerical solution converges to the origin but the true solution diverges to infinity. This phenomenon is named the abnormal convergence of numerical solutions. Abnormal convergence has not been studied fully; however, many concepts involving A‐stability have been established to treat the phenomenon that the numerical solution diverges even though the true one converges, as is apt to occur in integration of a stiff system.

An example will be illustrated here in which the standard Runge‐Kutta method generates abnormal convergence. Further, it will be proved that the abnormality is caused by the existence of an intersection of the absolute stability region and the right‐half plane. Even if the 1/8 formula method is used instead of the standard one, the same phenomenon can be seen because all four‐stage, fourth‐order, explicit Runge‐Kutta methods have the unique absolute stability region.

This paper seeks for Runge‐Kutta methods that are A‐stable and never generate abnormal convergence, even though this might be an excessive requirement. Concretely, two types of methods will be proposed whose absolute stability region coincides with the left‐half plane. Furthermore, it will be reported that Gauss‐Legendre methods are typical ones.
A method of realizing an adaptive normalized ARMA lattice filter
Authors: Haseyama Miki , Nagai Nobuo , Miki Nobuhiro
Published: July 2007

This paper proposes a method of realizing an adaptive normalized ARMA lattice filter. This filter was proposed in a previous paper and can be realized with an arbitrary ARMA order and with all the lattice coefficients less than one. Furthermore, these coefficients are equal, so the structure is simplified. The adaptive realization method proposed in this paper retains these features. In addition, we perform spectral estimation of a time‐varying signal with frequency weighting and model identification of an unknown input system. In our paper we describe an experiment with a real speech signal to prove that our algorithm can be used for frequency‐weighted spectral estimation. The use of real data to perform spectral estimation has not been attempted in other references.
An extension of the adaptive robbins‐monro procedure for a least‐square approximation problem of unknown systems
Authors: Maeda Yutaka , Kawamura Yoshiaki
Published: July 2007

This paper considers an unknown nonlinear static system with an adjustable parameter (input parameter). The problem is to find the least‐square approximation input parameter. A recursive algorithm is proposed that estimates this least‐square approximation parameter without an identification of the system. An output of the system is observed under a noisy environment. Suppose that the dimension of the output is equal to that of the input. In this case, the system characteristic can be regarded as a regression function, hence a recursive algorithm that can be constructed that minimizes the difference between the expectation of the output and a desired value of the output by using the stochastic approximation method or the adaptive Robbins‐Monro procedure. On the other hand, if the dimension of the output is greater than that of the input, the regression equation has no solution. Therefore, the usual stochastic approximation method or the adaptive Robbins‐Monro procedure cannot be used.

The algorithm proposed in this paper estimates the least‐square approximation input parameter that minimizes the squared error of the output. Especially, this algorithm agrees with the usual adaptive Robbins‐Monro procedure if the dimension of the output is equal to that of the input. Some numerical simulations are shown and the availability of the algorithm presented herein is confirmed.
A transistor‐island growing algorithm in a CMOS cell pattern generation
Authors: Shiraishi Yoichi , Sakemi Jun'Ya , Kutsuwada Makoto , Tsukizoe Akira
Published: July 2007

A transistor placement algorithm which generates a placement from a circuit diagram is described and evaluation results are presented. This algorithm is not guaranteed to be optimum; however, it can generate efficient placements for a wide range of cells with an equal or an unequal number of PMOS and NMOS transistors. The basic idea is to divide a problem into four subproblems, each of which independently attempts to minimize the number of diffusion separations, to straighten the gate‐wirings between PMOS and NMOS transistors, to minimize the estimated total wiring length, and to minimize the trunk density by utilizing the logical structure of an input circuit diagram. This algorithm was applied to the cell library generation and also to some “difficult examples” suggested in other papers. The generated cell patterns can actually be used and are comparable to other results.
Improvement on the error performance of M‐Ary PSK under a rician fading using time domain TCT
Authors: Li Huan‐Bang , Iwanami Yasunori , Ikeda Tetsuo
Published: July 2007

To reduce irreducible error floors that disturb digital mobile satellite links, a pilot tone‐based technique (TCT) has been presented in recent years. Although TCT performs well against the error floors, it is necessary for the TCT to make a null point at the transmitted data spectrum for carrying a pilot tone. This spectrum zero itself, however, usually degrades error performance. Further, the implementation of a narrowband pilot‐extracting BPF at the receiver is difficult. This paper presents a time domain TCT which can overcome these defects. Theoretical analysis and numerical results are given which verify the effectiveness of the presented system.
Design knowledge representation scheme for D‐A converter
Authors: Hayashi Shin‐Ichi , Watanabe Toshinori , Oosawa Hideyuki , Maio Kenji
Published: July 2007

With the advent of the application‐specific IC (ASIC) era, the diverse, small‐quantity production is required. With the progress of semiconductor integration, the fabrication is now in the era of the system‐on‐chip. Thus, it has become important to reduce the design period for the mixed analog/digital LSI. Compared to the digital circuit, the design techniques and the methods in CAD tools on analog circuits are still in the embryonic stage. The design of the analog circuit requires sophisticated decisions, such as the adequate selection from a very large number of possibilities for the topology of the circuit and the circuit constants, according to the purpose of application. Hence, the goal cannot be achieved by the traditional approach, where a relatively fixed problem‐solving procedure is formulated into a program along the particular algorithm. The approach by the knowledge processing will be useful.

This paper considers the design knowledge for the D‐A converter. A representation scheme for the design knowledge is proposed, whereby the design knowledge is described by being divided into a static part (production rule for the circuit) and a dynamic part (method of rule application). In this way, the design knowledgebase can be constructed efficiently. The proposed scheme actually is applied to the design of the D‐A converter, and it is shown that the scheme is useful in the design‐support, which has been difficult using earlier approaches. The experiment is made using Prolog as the knowledge processing language, and it is shown that the unification function and the backtrack function of Prolog is suited to the realization of the following design‐support functions for the analog circuits: (1) modification of circuit; (2) generation of abbreviated circuit; (3) documentation; and (4) net‐listing.
A method of constructing circuits containing several kinds of variable elements
Authors: Kawakami Atsushi
Published: July 2007

This paper considers the transfer function containing several kinds of variable parameters and proposes a method for synthesizing the circuit by the state‐space method. The method is to construct separately the denominator and the numerator of the given multiparameter transfer function.

Using the proposed method, the multiparameter transfer function can be constructed by the active RC circuit with the minimal number of dynamical elements and about half the number of variable elements, compared with the traditional method. A method is also shown which utilizes free parameters of the circuit structure to minimize the pole‐zero sensitivity of the multi‐parameter transfer function with respect to the variation of the active and passive elements. Finally, an illustrative example is shown to demonstrate the effectiveness of the synthesis method proposed in this paper.