bluuebunny/crossref_metadata_2025 huggingface dataset:
Active R filter realization and its trimming technique
Authors: Hayahara Etsuro , Hiranos Satoshi
Published: July 2007

This paper describes a design approach for constructing active R filters to simulate the doubly resistively terminated LC circuit and its compensation techniques. Using this approach, the necessary compensation resistance values can be obtained from only one measurement of the filter response.

The approach in this paper is applied to an all‐pole lowpass filter. However, it is shown that it can also be applied to a low‐pass filter having transmission zeroes.
Development of an educational computer simulator
Authors: Tsuchiyama Makio
Published: July 2007

The computer today is a tool that can be used by everyone. On the other hand, it is important for the student of information sciences to correctly understand the operational principle of the computer as a basic knowledge required in understanding or solving the problems concerning the computer.

To answer questions concerning the internal operation of the computer, it is necessary to understand the behavior of the actual circuit at the IC level. With the widespread use of the personal computer, it now seems possible to construct an efficient simulator for educational purposes. However, few reports have been presented concerning the approach at such a level.

From such a viewpoint, the author developed a computer simulator aiming at the understanding of the operation of the registers and gates based on the machine language program. Two modes, i.e., the hardwired logic and microprogram, are executed by the control unit of the system. The operation can be switched to the set‐up constructed by a student. Several elaborations are made so that the operation actually is felt. For example, the operation is represented by a time‐chart so that the time‐relations among signals can be understood.

A scheme is adopted whereby the generation program for the control signals is constructed directly from the time‐chart. It is expected that an efficient training for the operational principle over a wide range can be realized on a personal computer using the proposed system, relating the system behavior to the detailed circuit operations from the viewpoints of both software and hardware.
Adaptive cepstral analysis—adaptive filtering based on cepstral representation
Authors: Tokuda Keiichi , Kobayashi Takao , Shiomoto Shoji , Imais Satoshi
Published: July 2007

The unbiased estimation of the log spectrum is a method which is a stricter formulation of the cepstral method from the viewpoint of the spectral estimation. In the unbiased estimation of the log spectrum, the model spectrum is represented by cepstral coefficients and a spectral criterion is minimized with respect to the cepstral coefficients. By introducing an instantaneous gradient estimate of the criterion in a manner similar to the LMS algorithm, we develop an adaptive cepstral analysis algorithm. The properties and the features of the algorithm are discussed. In the analysis system, an IIR adaptive filter whose coefficients are given by the cepstral coefficients is realized by the LMA (log amplitude approximation) filter. The filter has an exponential transfer function and its stability is guaranteed.

In contrast to the conventional LMS algorithm, where the coefficients are updated based on the input vector and the output of the filter, the feature of the proposed filter is that the coefficients are updated based only on the output vector. The computational complexity in the update of coefficients and LMA filtering is of the order of0(M), whereMis the order of the analysis. The convergence property of the proposed algorithm is shown through the analysis of synthesized signals, and the usefulness of the method is demonstrated through an application example to the analysis of the natural speech.
A statistical modeling of charge redistribution a‐d converters for yield optimum design
Authors: Tateishi Tetsuo , Asadas Kunihiro
Published: July 2007

An analytical method has been applied to a charge redistribution A‐D converter. In this method, the element values of each constituent element in a circuit are treated as statistical probability variables. Hence, the probability distribution of the circuit characteristics of design objectives is derived analytically from the probability distribution of the element values. If the circuit characteristics are described in terms of the circuit element values, the probability distribution of the circuit characteristics is derived from the sensitivity analysis of the analytical expression.

In the case where the probability distribution of the circuit characteristics cannot be determined by the sensitivity analysis alone such as in the case of the error characteristics of binary weighted capacitor array, the characteristics are shown to be derivable as the probability distribution by combining the combinatory logic error analysis.

By means of the Monte Carlo simulation, the accuracy of the derived circuit characteristics is verified. By means of the probability distribution formula of the derived circuit characteristics, a formulation is carried out for design of the A‐D converter including the yield. A guideline for design by the nonlinear optimization is presented for the A‐D converter including the yield equation.
A hierarchical approach to module placement using neural networks
Authors: Shimamoto Takashi , Sakamotos Akio , Satoh Akiyoshi
Published: July 2007

A neural network model for solving optimization problems proposed by J. J. Hopfield et al. has yielded good solutions of the traveling salesman problem. The present paper describes a method of solving module placement problems in automatic LSI device layout, in which a neural network model is used to optimize a set quadrisection procedure by minimizing its objective function. The solutions obtained for this method are comparable to or better than those obtained by the GFDR method, an iterative‐improvement method of solving the module placement problem.
Characteristic extraction of speech using arma model with double inputs
Authors: Horita Eisuke , Miyanaga Yoshikazu , Tochinais Koji
Published: July 2007

This paper proposes a method of speech analysis using a new stochastic model. This model represents two different kinds of MA parts associated with pseudo‐periodical pulse train input and Gaussian process input. Therefore, the new model describes individually two different production models for voiced and unvoiced sound in the frequency domain and can precisely realize speech production models in comparison with an ARMA model. It has been shown from experimental results that the proposed stochastic model can extract more accurate characteristics of real speech than an ARMA model.
An estimation method of the optimal pairing and ordering for state‐space digital filters
Authors: Yoshikawa Toshinori , Wu Jun‐Hai , Oguros Yoshinari
Published: July 2007

This paper presents a method to estimate the optimal pairing and ordering method for cascade‐type state‐space digital filters, each having a second‐order section realized in 2D form or optimal form.

The optimality criterion is based on minimizing the normalized round‐off noise. The scaling method used is the L2 norm or L∞ norm. To show the effectiveness of the proposed method, this method is compared with four conventional methods. As numerical examples, the estimation method is shown for three filters having different orders.

The result of these examples allows a more optimal filter to be selected in terms of the optimality criterion of normalized round‐off noise when compared to the other conventional methods. Further, the proposed method also is capable of providing an estimation method for filters having section optimal form, of which the conventional methods are not capable. Finally, if the coefficient matrix is transformed using an equivalent transformation, it is also possible to extend the method to realizations other than 2D or the section optimal form.
Various structures of orthogonal projection‐type associative networks
Authors: Matsuoka Kiyotoshi
Published: July 2007

A typical model of associative memory is one based on the orthogonal projection operator calculated from the prototype patterns to be stored in a network. This paper shows that the orthogonal projection‐type association can be realized by various structures of neural networks. Although they are almost equivalent from a mathematical viewpoint, each has particular characteristics which are useful from some practical points of view, particularly when implemented in hardware.
Decomposition of three‐terminal nets in two‐layer routing of PC boards
Authors: Masuda Sumio , Kashiwabara Toshinobu , Fujisawa(Dec.)s Toshio
Published: July 2007

Suppose that terminals are located on a simple closed curve. For a case where only the inside area of the closed curve is the wiring region and there are two layers available for wiring, we consider a problem of finding a realization, if any, of a given routing requirement (set of nets) with no throughholes.

It is known that if each net contains only two terminals, this problem is easily solved. Thus in this paper, an algorithm is proposed for the case when each net contains no more than three terminals. The time complexity of the algorithm herein is 0(n2), where n is the number of nets.
Noise analysis method for nonlinear circuits with two frequency excitations using the computer
Authors: Okumura Makiko , Tanimoto Hiroshi , Sugawaras Tsutomu
Published: July 2007

This paper considers the noise in the nonlinear circuit, which operates by two signals i.e., the input signal and the periodic excitation (clock signal or carrier signal). These include the switched‐capacitor circuit and the mixer circuit. A method for the numerical analysis of the noise by a computer is proposed. The aliasing of the frequency component arises in such circuits, and the aliasing noise component over a wide frequency range must be considered in the noise analysis.

In the proposed method, among the two excitations, the input signal is assumed as very small. The nonlinear circuit with only the periodic excitation is linearized around the periodic solution. Then a periodically time‐varying circuit is derived. An equivalent current source is connected in parallel with each noise source and its output noise is computed. Aliasing components folded back to the baseband from high‐frequency bands are calculated, and their powers are accumulated until their contributions become negligible.

The proposed method is built into the circuit simulator SPREAD [1] and an experiment was executed. The usefulness of the algorithm was verified using a simple circuit, for which the strict solution can be calculated analytically. By the proposed method, the noise of the nonlinear circuit with two frequency excitations can be estimated at the design stage. This will greatly improve the productivity in the circuit design.
A note on connection between spline signal spaces and band‐limited signal spaces
Authors: Toraichi Kazuo , Kamadas Masaru
Published: July 2007

If the signal space matched to the representation is selected according to the properties of the signal to be handled, the signal processing can be made efficient. This paper is a step toward the structuralization of the signal spaces for systematic selection of the signal space. As a particular problem, the relation is analyzed between the signal spaces composed of spline functions, which have the feature that the smoothness as the continuous differentiability varies according to the degree, and the band‐limited signal spaces, which is fundamental in the signal processing.

As a result of analysis, it is shown that the signal spaces composed of spline functions agree with the band‐limited signal spaces in the limit of infinite order. This implies that the signal spaces composed of spline functions and the band‐limited signal spaces can be structured as a sequence with the degree as the parameter. The result in this paper provides a view that the continuous differentiability can be used as an index in the systematic selection of the signal spaces.
A cyclic ∑Δ ADC using time‐sharing technique
Authors: Chigusa Yasutami , Tanaka Mamoru
Published: July 2007

A conventional ∑Δ ADC has superior properties with its extremely simple construction, its robust instrumentation for various parasitic effects, and so on. However, the conversion rate is notably low. This paper proposes faster and higher‐resolution ∑Δ ADC with an additional analog feedback line which enables the time‐sharing technique to be employed. Changing the system clocks can control the rate of the quantization errors and the conversion time of the novel cyclic ∑Δ ADC. This study focuses on the area between a sampled ∑Δ ADC and a successive approximation converter. The sufficient results of simulation and experiments were obtained using discrete devices where the performance of the cyclic converter was compared with that of the conventional one.
A comparison of synthetic speech quality generated by the residual‐and multipulse‐excited analysis‐synthesis methods
Authors: Takedas Shoichi , Asakawas Yoshiaki , Ichikawas Akira
Published: July 2007

A source coding technique using multiple excitation pulses is expected to improve the quality of rule‐based synthetic speech.

In this paper, residual‐ and multipulse‐excited analysis‐synthesis methods are compared by evaluating the relationship between speech quality and bit rates. A method is proposed for reducing bit rates by thinning out prediction residual pulses (Thinned‐Out Residual method).

At excitation pulse utilization rates greater than 50 percent, this residual‐excited analysis synthesis method produces higher‐quality (segmental SNR) speech than the multipulse‐excited analysis‐synthesis method. When high‐quality synthetic speech is required, the optimal residual utilization rate therefore is 100 percent. When a low bit rate is required, however, the pulse number should be reduced greatly. The multipulse‐excited analysis‐synthesis method then synthesizes speech with a higher segmental SNR.
Considerations on irreducible cyclotomic polynomials
Authors: Sugimura Tatsuo , Suetugu Yasunori
Published: July 2007

Originally irreducible polynomials were obtained essentially by checking the irreducibility of a randomly chosen polynomial. Recently, irreducible polynomials of a specified form have been produced and a transformation of the variable has been applied to produce a systematic method of deriving irreducible polynomials. In this paper, we consider irreducible polynomials from the viewpoint of cyclotomic polynomials. We define conditions for the existence of irreducible cyclotomic polynomials and show that these polynomials have some specified forms. Additionally, it is shown that an infinite number of irreducible polynomials can be generated which satisfy those conditions.
Extended‐order statistic filter and evaluation of noise reduction performance
Authors: Fujii Tatsuya , Arakawa Kaoru , Harashima Hiroshi
Published: July 2007

The authors investigated construction of an extended‐order statistic filter to remove non‐Gaussian white noise added to image signal. First, they devised a differential‐order statistic filter using signal‐differential ordering instead of a signal‐value ordering and estimated its noise reduction performance. Next, they proposed an extended‐order statistic filter which unifies the three‐order information of time, signal value, and signal differences obtained from a signal within the filter window.

Preparing the 3‐dimensionalized filter coefficient space and using filter coefficients chosen sequentially in the three different orders among them allow effective noise reduction performance to be obtained. In addition, as a method for resolving a problem on the design method of filter coefficients, investigation was made using a synthesized training sequence. Extracting parameters necessary to a linear AR model from an objective signal, it is shown that the filter design using a training sequence synthesized from it gives noise reduction performance similar to the one at the optimal design.
Sound field analysis of ultrasonic probe with many medium boundaries
Authors: Okada Hisao
Published: July 2007

For the field analysis of an ultrasonic probe, the method based on Huygens' principle often is employed. Most such calculations, however, are concerned with the cases where the sound source and the region of calculation belong to the same medium or belong to a different medium divided by a single planar boundary.

This paper reports on the two‐dimensional method of field analysis, which is based on Huygens' principle, but can be applied even if a large number of curved medium boundaries exists between the source and the region of calculation. The method analyzes the field as follows: the vibration amplitude and transmission coefficient on the medium boundary closest to the source are calculated. The product of the vibration amplitude and the transmission coefficient is calculated, and the medium boundary is regarded as the secondary source with the product value as the vibration amplitude. Then the field at the next medium boundary or in the next medium is calculated.

The medium boundary is divided into line elements, and the transmission coefficient is calculated by determining the angle of incidence from the phase difference of the vibration at both ends of the element. As an application example of the proposed method of field analysis, the focused field is analyzed which is formed by the acoustic lens, cylindrical vibrator and the phased array.
Spectral efficiency of M‐ary/spread spectrum multiple access communication systems
Authors: Tachikawa Shin‐Ichi , Marubayashi Gen
Published: July 2007

In this paper, M‐ary/spread spectrum multiple access communication systems (SSMA) are studied to improve spectral efficiency. A novel method of calculation of the spectral efficiency for several tip waveforms and bit error rate is shown. For M‐ary/ SSMA using orthogonal codes, the spectral efficiency can be improved considerably, however, the code length is becoming too long to achieve satisfactory spectral efficiency. To shorten the code length for the same spectral efficiency, M‐ary/SSMA is adopted using concatenated Reed‐Solomon codes, and the effectiveness is shown.
A construction of M‐channel filter banks with hilbert transformers
Authors: Wada Shigeo , Takahashi Shin‐Ichi
Published: July 2007

This paper proposes a new design method for the filter bank. The filter bank is a system which can reconstruct a signal via band‐splitting, decimation, interpolation and restoration. It is employed widely to solve various problems in the communication system, and various methods have been presented for its specification, implementation and design.

This paper discusses the approximate realization of the filter bank, which is called the M‐channel uniform‐splitting FIR perfect‐reconstruction filter bank, and its design principle is as follows. First, the analysis filter is designed so that each channel has the characteristic which is the translation of the given LPF on the frequency axis. Then the synthesis filter is designed in the time‐domain so that the unit impulse response of the whole system is equivalent to the unit pulse, i.e., that the response of the linear‐phase all‐pass transmission characteristic. The Hilbert transformer plays the important role in the translation of the characteristic. Compared with the conventional design, the proposed method has the merit that no constraint is imposed on the band splitting and the band is partitioned uniformly with an arbitrary number of divisions. Design examples are shown to indicate the usefulness of the proposed design.
Construction of 2‐D digital filters using all‐pass networks with complex coefficients
Authors: Toyoshima Hisamichi , Takahashi Shin‐Ichi
Published: July 2007

Several methods have been presented recently for constructing the one‐dimensional IIR digital filter by combining all‐pass networks. It is known that the filter composed of the all‐pass networks has a better computational efficiency and also low coefficient sensitivity due to the structural boundaries. In addition to the one‐dimensional filters, a construction of the two‐dimensional filter based on the all‐pass network has also been proposed. A problem in that method, however, is the restriction on the realizable characteristics.

This paper attempts to overcome that problem, and proposes a new construction of the two‐dimensional digital filter using the all‐pass network. The new approach employs the complex coefficient all‐pass network instead of the conventional real‐coefficient all‐pass network. By this approach, the degree of freedom in the design is enhanced, and the characteristics which have been difficult or impossible to realize in the case of the real coefficient filters, are now made realizable.
Two‐dimensional analysis of coupled thickness‐shear and flexural vibrations in rectangular AT‐cut quartz resonators using a finite‐element method
Authors: Sekimotoy Hitoshi , Watanabe Yasuaki , Tanaka Keisuke , Nakazawa Mitsuo
Published: July 2007

This paper proposes a method of two dimensional analysis for the thickness‐flexural vibration, where the one‐dimensional finite‐element method (FEM) is applied to the planar equation. This approach aims at application to the analysis of the flexural spurious vibrations of strip rectangular AT‐cut plate often used in practice.

The feature of the proposed method is that the two‐dimensional coupled vibration can be analyzed by applying twice the one‐dimensional FEM. In other words, the one‐dimensional FEM is applied to the eigenmode guided by the X‐edge, as well as to the resonant expression derived from the boundary condition on the Z'‐edge. As a result, the considered matrix is reduced greatly in size compared to the direct analysis by the two‐dimensional FEM. The result of calculation for the change of the resonant frequency due to the change of the Z'‐edge ratio is compared to the result of experiment, and the usefulness of the proposed method is demonstrated. It is shown also that most of the spurious vibrations depending on the Z'‐edge ratio are flexural vibrations, which is a new finding.
A design method of window functions by spline approximation
Authors: Toraichi Kazuo , Kamada Masaru , Kawamoto Ryuichi , Ishii Rokuya
Published: July 2007

In a previous paper, the authors proposed a design method for representing the window function by the C‐spline function, which is a linear combination of the B‐spline functions. By using that design method, a window function is obtained which has the same mainlobe bandwidth and the highest sidelobe level as the Hanning window or Blackman window, but a faster sidelobe fall‐off. However, the design method for the new window function is not practical due to the fact that the description of the design specification is premature and the optimal dimension is not specified.

This paper aims at the realization of a practical design method by improving those points. A design method is proposed in which the specification for the high‐frequency range is given in terms of the sidelobe fall‐off, and the specification for the low‐frequency range is given in terms of the desired frequency characteristics in the logarithmic scale, together with the tolerable approximation error, represented by the mean‐square error.

The optimal dimension is determined, in the iterative approximation increasing the dimension to reduce the error in regard to the desired frequency characteristics, as the minimum dimension that reduces the error to the tolerable value. The usefulness of the proposed design method is demonstrated by a design example for the window function, which has a better characteristic than those of Kaiser‐Bessel, Gauss and Hanning windows.
On detection of quantization noise in low level reproduction of digital audio system
Authors: Ebata Masanao , Maeda Kanako , Takada Kenji , Usagawa Tsuyoshi
Published: July 2007

On recent audio equipment, a wide dynamic range of nearly 100 dB is achieved by using 16 bit linear quantization. But quantization noise is still detectable, because relatively few bits are used to represent low‐level signals. When the signal is quantized directly, the quantization noise consists of harmonics of the source frequency. This quantization noise is detectable at 55 dB for a pure tone and at just about 40 dB for complex and modulated tones, if we assume that the maximum reproduction level is 120 dB. This harmonic distortion can usually be reduced by adding dither at quantization. When dither is added, the quantization noise becomes white noise and spreads over the entire frequency range. Experimental results indicate that at frequencies lower than 2 kHz, the detection level of quantization noise is raised by several dB when dither is introduced. But at about 4 kHz, the detection level falls because of the dither; in this case, the white noise induced by the dither seems to function as a detection cue.
Pole assignment for three‐dimensional systems using two‐dimensional dynamic compensators
Authors: Kawakami Atsushi
Published: July 2007

This paper discusses the pole assignment problem for the three‐dimensional system which is described by a state‐space model. The problem is to provide a feedback and input transformation using two‐dimensional dynamic compensators to the given three‐dimensional system. In this way, the denominator polynomial of the transfer function of the closed‐loop system coincides with the product of three one‐dimensional polynomials with roots in the unit circle.

First, the three‐dimensional system is considered as a one‐dimensional one over a two‐dimensional rational functional field, and the desired pole assignment is realized by providing the two‐dimensional dynamic feedback and the input transformation. Concretely, the dynamic compensators for the feedbak system and the input transformation are regarded as two‐dimensional transfer functions. By providing the minimal realizations of those in two steps, the feedback system and the input transformation system are realized. Then, a detailed discussion is given to the realizability of those two‐dimensional dynamic compensators, i.e., the possibility that the dynamic compensators are proper rational functions with respect to both variables.

A counter‐measure for the nonproper case is also proposed. A condition for the obtained three‐dimensional system to be stable is derived. The dimension needed in the realization of those two‐dimensional dynamic compensators, i.e., the dynamical dimension needed in the pole assignment, is calculated. A method for realizing the pole assignment with the lowest dynamical dimension is discussed.
A variable step gain algorithm with fuzzy control
Authors: Katoh Akil , Kubota Hajime
Published: July 2007

Numerous methods for a parameter estimation in adaptive signal processing have been proposed. LMS, learning identification method, etc., for example, often are used due to easiness of their hardware construction, but their estimation speed and estimation accuracy vary by contrast depending on the magnitude of the coefficient called step gain. As a method for resolving it, a variable step (VS) algorithm in which the step gain is changed according to the parameter estimation condition has been proposed. However, since the variables in these algorithms that are intended to express the estimation conditions do not fully do so, it remains difficult to control the step gain appropriately.

This paper defines anew variables to detect the change of parameters and disturbance and to comprehend the parameter estimating condition, and proposes a variable step algorithm for controlling the step gain by fuzzy controller with these variables as fuzzy input. Furthermore, computer simulation is implemented by applying this method to the learning identification method to show its effectiveness. In particular, with this algorithm we do not need to change the algorithm to deal with variations of the statistical properties of the input signal, and the computational complexity does not depend on the order of the filter; thus it is advantageous for higher‐order identification. Moreover, we have observed that it can cope with observation of noise, rapid change of unknown parameter, and it improves estimation speed by about 15 percent and estimation accuracy by about 20 percent compared with those of existing VS algorithms.
Equivalence of hidden Markov models
Authors: Kabayashi Kingo , Amari Shun‐Ichi , Ito Hisashi
Published: July 2007

The hidden Markov information source (process) is a stochastic process with a finite‐state Markov chain behind it, and the state cannot directly be observed, while only a function of the state is observed as a stream of symbols. This kind of process is very important in both theory and application, but its theoretical structure has not been clarified.

This paper gives a sufficient condition for two hidden Markov processes based on different Markov chains to be equivalent as the stochastic process. The condition for that condition to be necessary also is shown. A condition for a hidden Markov process to be equivalent to a Markov chain also is presented.
Evaluation of human exposure to lon‐current using boundary element method
Authors: Yamashita Masaji , Hatori Kozo , Shimizu Koichi , Matsumoto Goro
Published: July 2007

With a view to evaluating the human exposure to ion current generated under a dc power transmission line, the exposure was evaluated by means of a numerical method (1) a new ion current analysis method was developed which can be applied to bodies with complex shapes such as biological bodies. (2) The method is based on the boundary element method, (3) the space charge terms were processed under an assumption of a lumped source of ion current, and (4) comparing the theoretical and experimental data, the error was evaluated. The method was then applied to biological bodies. It was found that the electric field on the surface of the biological body in the ion current field becomes larger than the one obtained in the Laplace field without ion current. This result indicates the importance of the analysis of the ion current field. Further, the variation of the electric field concentration accompanied by changes in the posture of the human body were clarified. The variation of the surface electric field when a body is charged by the ion current was clarified, as well. Hence, this paper provides useful information on the determination of the safety standard and on evaluation of the biological effects of ion current field.
Speech recognition using VQ‐codebooks representing hierarchical spectral dynamics
Authors: Furui Sadaoki
Published: July 2007

This paper proposes recognition methods that use hierarchical spectral dynamic features extracted over multiple time lengths. The effectiveness of these features in phoneme recognition and isolated word recognition is shown. The dynamic features are extracted for time sequences of both cepstral coefficients and logarithmic energy, and are combined with instantaneous cepstral coefficients. Input speech is quantized by word‐or phoneme‐specific codebooks created by clustering a set of vectors consisting of these features for each word or phoneme.

Speaker‐independent isolated word recognition experiments are performed using a vocabulary of 100 Japanese words. When VQ distortion is used for word identification, the high recognition accuracy of 96 percent is achieved, and when VQ distortion is used for preprocessing, the number of word candidates for each input utterance is reduced to 1 percent of the vocabulary without increasing the error rate. Phoneme recognition experiments are performed for the /b/, /d/ and /g/ consonants in a large vocabulary of isolated words uttered by one male speaker. Using the proposed recognition method, the high recognition accuracy of 99 percent, which is similar to the accuracies obtained by conventional HMM or neural network approaches, is obtained. This paper also compares multiple codebook and single code‐book methods.
Constructive analysis for infinite‐dimensional nonlinear systems—infinite‐dimensional version of homotopy method
Authors: Makino Mitsunori , Oishi Shin'Ichi
Published: July 2007

Related to the problem such as determination of the operating point for the nonlinear resistor circuit equation, remarkable progress has recently been observed in the global solution method for the finite‐dimensional nonlinear equation, called the homotopy method.

This paper attempts to extend the homo‐topy method to the infinite dimensional nonlinear equation (functional equation) such as the equation for the semiconductor devices. First, two streams are pointed out as the theoretical frameworks for the homotopy method for the finite‐dimensional equation, which are: (1) the characterization of the solution structure for the nonlinear equation; and (2) the tracing of the solution trajectory. It is then pointed out that the Fredholm operator equation can be considered as a class of infinite‐dimensional equations, which can characterize the set of solutions as in item (1).

The existence theorem for the solution in the finite‐dimensional homotopy method is extended to the class of Fredholm operator equations. Then it is pointed out that the A‐proper operator equation can be considered as a class of infinite‐dimensional equations, for which the computational algorithm for the solution can be described as in item (2). Based on the constructive theorem for the existence of the solution in the finite‐dimensional homotopy method, it is shown that the solution for the particular class of A‐proper operator equations can always be determined by numerical calculation. Finally, an application to the controllability problem is discussed.
A method to minimize output noise in cascade SC filters
Authors: Ma Zhiqiang , Nakayama Kenji , Yamamoto Gaishi
Published: July 2007

The output noise in the cascade switch‐ed‐capacitor (SC) filter depends greatly on the pole‐zero combinations as well as the order of cascading the second‐order circuits. This paper derives the general condition for the combination/order so that the output noise is minimized.

First, the noise transmission from each noise source to the output is analyzed for the second‐order circuit. It is shown that the sample‐and‐hold component in the output noise is sufficiently large compared to the continuous‐time component, and the effect of the pole‐zero combination is small.

Then, the condition for minimizing the noise transmitted from each second‐order circuit to the output of the SC filter is investigated. The derived condition is that the amplitude response of the transfer function from the filter input to each second‐order circuit should be flat. Translating the condition into the condition for the pole‐zero combination/cascading order, the following condition is derived. For the combination, starting from the pole with higher Q, the closest zero is combined; for the order, the circuits are cascaded so that the frequency at which the amplitude response from the filter input to the second‐order circuit should be distributed.

Finally, simulations are executed for 11th‐order low‐pass and high‐pass filters. It is seen that for approximately 40 percent of 36 ways of combination/order with the foregoing condition, the output noise ranges within 20 percent of the minimum value, which indicates the effectiveness of the presented condition.
Relaxation‐based spectrum analysis for nonlinear circuits
Authors: Kawashima Takahiro , Higashiyama Takeshi , Asai Hideki
Published: July 2007

In the spectrum analysis of a nonlinear circuit by the conventional harmonic balance method, a problem occurs when a large number of harmonic components is assumed, that is, the Jacobian matrix of Newton's method for the solution of the determining equation has a large dimension which requires a long computation time and a large memory capacity.

From such a viewpoint, this paper proposes an algorithm for the spectrum analysis in the harmonic balance method, where the nonlinear relaxation is applied to the solution of the determining equation. A continuous method is introduced to cope with the convergence problem of the harmonic relaxation method in the strongly nonlinear circuit. Then, to handle the multinode circuit, an iterated spectrum analysis is proposed in which the harmonic balance is applied to each node individually. It is shown that the computation speed in the analysis is improved greatly, while drastically saving the memory capacity in the proposed method, compared to the conventional method based on Newton's method.
Synthesis of an LC filter suitable for LC simulation‐type RC active filters and its application
Authors: Ishibashi Yukio
Published: July 2007

One of the most important measures for the construction of the active RC filter is the element sensitivity. One of the effective means to reduce the element sensitivity is the method to simulate the conventional LC filter with both ends terminated by resistors.

This paper considers the realization of the low‐pass filter by LC simulation using GIC or by the node‐voltage simulation. The prototype LC filter is discussed to solve the bias instability problem, which is a problem in the realization.

First, an LC filter is proposed which is obtained by adding an inductor in parallel with the input resistor of the conventional doubly terminated LC filter. The method of constructing the filter is discussed. Then a method is presented to simulate the filter using GIC. Finally, the 5th‐order Chebyshev low‐pass filter is constructed and its characteristics are measured. The element sensitivity is analyzed using a computer, and it is verified that the proposed circuit has a low element sensitivity.
A clustering algorithm to minimize recognition error functions
Authors: Ando Akio , Ozeki Kazuhiko
Published: July 2007

Pattern recognition methods in which multiple templates are prepared for each category and which recognize input patterns using those templates usually partition training patterns into clusters and generate templates as centroids of each cluster. However, most clustering algorithms proposed in the literature aim at minimizing the square error distortion measure when each cluster is represented by its centroid. Therefore the result obtained is not necessarily optimal from the viewpoint of discrimination.

This paper describes a new clustering algorithm which is a method of obtaining an optimum solution to a template generation problem. The simulated annealing is used for searching for a cluster partition that minimizes the “degree” of recognition error when all training patterns are recognized by using centroids as templates. This paper also describes a vowel template generation problem in speech recognition as an example of pattern discrimination problems. The new algorithm yielded better experimental results than the LBG algorithm and the LVQ algorithm.
Maximum entropy sampling frequency for stationary gaussian signals
Authors: Sakamoto Hiroyasu
Published: July 2007

In various kinds of digital signal and information systems, there are strong relations between the sampling frequency of signals and the performance or efficiency of the systems. Therefore, many reports have been made on optimization of sampling frequency for various application systems.

This paper discusses the problem of determining the optimum sampling frequency for stationary Gaussian signals which does not necessarily satisfy the band‐limited condition of the sampling theorem.

First, a brief survey of conventional decision criteria is presented. Based on a common understanding for the amount of information of a sampled datum, a new criterion is proposed that maximizes the entropy rate per one sample. For stationary Gaussian signals, it is expressed in power spectral density functions accompanied by sampling.

The criterion compensates for the following defect of the conventional auto‐entropy rate by subtracting the cross‐entropy rate: not only does it properly reflect loss of information due to the occurrence of spectral aliasing error, but also it estimates the spectral error to be information gain.

Several properties of the criterion are obtained. As numerical examples, the last part of the paper shows methods for solving the equations for the optimum sampling frequency and their solutions for signals which have nonoscillatory or oscillatory exponential auto‐correlation functions.
An algebraic analysis for bifurcation problems
Authors: Fujisaka Hisato , Sato Chikara
Published: July 2007

An algorithm for computing the codimensions of tangent space for bifurcation equations using the Gröbner basis is presented. When a bifurcation equation with perturbation added becomes equivalent to the original bifurcation problem, the perturbation term belongs to the tangent space; this contains an ideal of a certain equation set on real functions. Therefore, perturbation terms that prevent the equivalence relationship belong to the codimensional space of the tangent space. On the other hand, the residue system of ideals on polynomial rings is found as a set of terms that are not multiples of the highest order of the Gröbner basis.

In this paper an algorithm will compute the codimension algebraically by associating the codimensional space of a tangent space with a residue system. Thus, a symbolic manipulation description is adequate for this algorithm. An established algorithm can be similarly applied not only to the case in which a bifurcation equation can be expressed by one variable and one parameter but also to the case in which it is expressed by simultaneous equations of multivariables and one parameter. As an example of computation an equation expressing bifurcation such as Pitch‐fork and Hilltop, etc., was used. The results were verified by illustrating the solution curve when perturbation terms belonging to the codimensional space are added.
A statistical analysis algorithm based on the continuation method
Authors: Inoue Yasuaki
Published: July 2007

Performance of analog IC is affected largely by variations in component characteristics. Therefore, it is necessary to verify at the design stage that the performance of a designed circuit sufficiently satisfies the specifications. Especially important is the verification of variations in dc operating points, which is the foundation of circuit operation. A circuit simulator with statistical analysis function by Monte Carlo method is known as a CAD tool for this verification.

In the statistical analysis of dc operating points, reiterative operations are required to obtain repeatedly a solution of a system of nonlinear equations. Thus, the convergence performance of a solution algorithm comes into question. Here, the continuation method to the convergence problem in a statistical analysis of dc operating points of analog ICs is applied. and it is shown that convergence rate and reliability can be greatly improved. First, the computation‐efficient Davidenko approach is shown to be applicable to the statistical analysis of dc operating points of analog ICs. Next, based on this approach, an executive algorithm, the statistical parameter stepping method (SPS method) is proposed. This algorithm improves convergence performance. Moreover, by applying this algorithm to statistical analysis of real analog ICs, its effectiveness is demonstrated.
Design theory of piezoelectric ceramic transformer as an asymmetric filter
Authors: Hirose Seiji , Shimizu Hiroshi
Published: July 2007

The piezoelectric ceramic transformers as a source for a high voltage is used in a high‐energy density state due to its small size so that the energy loss causes increased heat generation and reduction of efficiency. Therefore, it must be treated not only for the voltage step up but as a power transmission system. In the conventional design method for the piezoelectric ceramic transformer, only the voltage ratio was considered important.

In this paper, the piezoelectric ceramic transformer is considered as an asymmetric filter with a different capacitance ratio and different termination resistances of the input and output parts. The transmission frequency and the termination condition for a low propagation loss and a high‐voltage ratio were found. As a figure of merit of the piezoelectric ceramic transformer, the voltage ratio x relative bandwidth was proposed. By taking into account this value and the transmission attenuation, an optimum circuit condition of the piezoelectric ceramic transformer was obtained.
Nonrecursive multiple filters on an integer residue ring
Authors: Murakami Hideo , Yamano Gosuke , Koyama Yoichi
Published: July 2007

The finite ring is used for efficient computation of filters. This paper discusses nonrecursive multiple filtering systems on an integer residue ring. The number theoretic transform, which is the Fourier transform on the finite ring, has been used for the design of FIR digital filter on an integer residue ring. A set of filters can be combined to a single filter on a finite ring, using the isomorphism derived from the Chinese remainder theorem. This equivalence is used to compute the outputs of the set of filters efficiently.

An efficient algorithm is also presented to find a primitive 2t‐th root of unity which is needed for the construction of the numbers theoretic transform. For filtering multiplicity ranging from two to five, possible choices of moduli are presented. Finally, an example and evaluations of the nonrecursive multiple filter are shown.
Network reliability analysis using dual graph
Authors: Hayashi Masahiro
Published: July 2007

The problem of representing a communication network in a graph and evaluating the overall reliability is NP‐hard. The class of graphs called planar CF graphs is the maximum class whose exact overall reliability is confirmed as being possible to be evaluated by the polynomial order.

In this paper, the dual graph to the given graph is constructed, and the overall reliability is evaluated. As a result, the following properties are obtained.

The evaluation of the overall reliability is equivalent to the evaluation of the probability that a circuit exists in its dual graph (dual problem).

The exact value of the dual problem can be evaluated by the reduction method.

There exists a class of graphs which permits the evaluation of the overall reliability by the polynomial order using the method of property (2), and is larger than the Planar CF Graph (we call this extended class Planar Dual CF Graph).
A piecewise‐linear homotopy method for nonlinear programming
Authors: Yamamura Kiyotaka , Arai Kaori , Kiyoi Masahiro
Published: July 2007

In nonlinear programming problems, an objective function f(x) is optimized (maximized or minimized) subject to some constraints. Such problems are also called constrained optimization problems. Most of the algorithms in nonlinear programming are classified into two categories: 1) transformation methods; and 2) projection methods.

The homotopy methods, which are the subject of this paper, belong to the category of projection methods. The main feature of the homotopy methods compared with other projection methods is that they are good at global convergence (which is lacking in most of the projection methods) but are not good at convergence speed (which is the strong point of most of the projection methods).

This paper discusses the homotopy methods in nonlinear programming and show that the piecewise‐linear homotopy method using the Newton homotopy and polyhedral subdivision is very effective for solving nonlinear optimization problems. A new algorithm is proposed that exploits the partial separability and linearity of the Kuhn‐Tucker equations (which appear in the nonlinear programming problems). By this exploitation, the computation efficiency is improved markedly compared with the conventional homotopy methods using simplicial subdivision. Moreover, the proposed algorithm converges quadratically, thus accurate solutions can be obtained rapidly. It is proved also that the proposed algorithm is globally convergent for the constrained convex optimization problems. Except for the shortcoming that the programming is complicated, the proposed algorithm has wellbalanced effectiveness.