bluuebunny/crossref_metadata_2025_split huggingface dataset:
Development of an educational computer simulator
Authors: Tsuchiyama Makio 
Published: July 2007

AbstractThe 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

AbstractThe 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

AbstractAn 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

AbstractA 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

AbstractThis 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

AbstractThis 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

AbstractA 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

AbstractSuppose 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

AbstractThis 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

AbstractIf 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

AbstractA 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

AbstractA 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

AbstractOriginally 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

AbstractThe 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

AbstractFor 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

AbstractIn 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

AbstractThis 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

AbstractSeveral 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

AbstractThis 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

AbstractIn 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.