bluuebunny/crossref_metadata_2025 huggingface dataset:
Resonance frequency analysis by galerkin's method for clamped‐free, stepped flexural mode quartz crystal resonators and its application to a quartz crystal tuning fork
Authors: Kawashima Hirofumi , Sunaga Kenji
Published: July 2007

This paper describes a stepped flexural mode quartz crystal resonator. The objective of this paper is to determine the resonant frequency, the motional series resistanceR1, and theQvalue of the stepped flexural mode tuning fork‐type quartz crystal resonator. The equation of motion is presented for a flexural mode resonator with nonuniform cross section with rotary inertia, and shear force is taken into consideration. Then, since the flexural mode quartz crystal resonator can be thought of as having two clamped‐free beams, the vibrational displacement computed from the boundary condition of clamped‐free ends and the condition of continuity of different cross sections are used to derive the frequency equation by Galerkin's method.

In addition, the transcendental equation can be derived from the boundary conditions and the continuity condition. Further, from these relationships, the frequency constant (f·l1) and the resonant frequency are determined as functions of the width ratioW21(widthw2/‐widthw1) and the length ratioL21(lengthl2/lengthl1). As a result, it is shown that the frequency can be lowered without increasing the length of the resonator. These relationships are compared with the experimental results. It is shown that both agree very well. Hence, the present analysis results are effective for arbitrary frequency design. Finally, the typical values of the electrical equivalent circuit parameters are given. It is shown that, asL21is increased, the motional series resistanceR1increases and theQvalue decreases, although the characteristics are sufficient as a resonator. At the same time, the structure is resistant to vibration.
Software operation system information model
Authors: Kobayashi Toru , Horikawa Keitaro , Ushida Fumiaki
Published: July 2007

A domain model for software maintenance and delivery is proposed and a specific information model for a telephone switching support system is derived from it by using object‐oriented analysis. This domain modeling and information‐model construction method can easily define boundaries for complicated tasks and can be reused for other target systems. Since the derived information model does not have any redundant objects or services, it is believed that it will facilitate the development of systems that have consistency among objects and specialized services.
Adaptive phase control using recursive least squares (RLS) phase estimation
Authors: Denno Satoshi , Saito Yoichi
Published: July 2007

This paper considers the realization of high‐speed phase synchronization when there exists a frequency error and discusses the application of the recursive least square (RLS) algorithm to the burst signal demodulator. Especially, as a phase control algorithm for the coherent detection where there is an error in the carrier frequency, a method is proposed to represent the phase variation by the frequency error as the state transition function and a detailed algorithm is derived.

It is shown that the theoretical convergence and the tracking performance of the algorithm agree with those of the ordinary RLS algorithm. The algorithm is applied to the frequency control circuit of the π/4‐QPSK differential detector and the phase control circuit of the 16‐QAM quasi‐coherent detector.

It is demonstrated experimentally that the system has an excellent performance and that the initial convergence is complete in approximately ten symbols when there exists a frequency error.
Weak radar signal detection based on wavelet transform
Authors: Ehara Naoki , Sasase Iwao , Mori Shinsaku
Published: July 2007

Weak signal detection is a basic and important problem in radar systems. The detectable radar range is determined by the radar equation, and radar performance can be improved by increasing the receiver output signal‐to‐noise ratio (SNR). The pulse integration method has been used to improve SNR. The new method proposed here, based on wavelet transform, has been used primarily for data compression in image processing. Since wavelet transform performs signal analysis using the wavelets that are localized in both time and frequency domains, better results are expected than would be expected if the conventional method of pulse integration in time domain is used. Using numerical simulations that assume white Gaussian noise is generated internally in the receiver, we show that an improvement of approximately 8 dB in SNR is obtained for only one pulse and high performance radar can be obtained using this method. This method can also be applied for signal detection in other digital communications.
Speaker adaptation using spectral interpolation for speech recognition
Authors: Shinoda Koichi , Iso Ken‐Ichi , Watanabe Takao
Published: July 2007

This paper proposes a technique for supervised speaker adaptation for speech recognition based on a continuous density hidden Markov model (HMM). When the number of utterances for the adaptation words is decreased in order to reduce the burden on the user, the ratio of demisyllables (recognition units) not appearing in the uttered words is increased. In the proposed method, the HMM parameters for the demisyllables contained in the adaptation data are corrected and then the HMM parameters for the demisyllables not contained in the adaptation data are corrected by interpolation in the parameter space (spectral interpolation). To avoid a biased estimation of the parameters that depend on the adaptation data set, a correction is made based on large‐scale uttered data by a large number of speakers. A word recognition experiment, executed for 100 similar words, simulates a 5000‐word large‐vocabulary speech recognition; and the proposed method is then evaluated. When the recognition rate by the speaker‐independent HMM reaches 81.2 percent, the recognition rate is improved up to 85.2 percent by using 50 words for adaptation. Thus, the effectiveness of the proposed method is verified.
Vector quantization of LSP parameters using moving average interframe prediction
Authors: Ohmuro Hitoshi , Moriya Takehiro , Mano Kazunori , Miki Satoshi
Published: July 2007

This paper proposes a new method for vector quantization (VQ) of LSP parameters, using moving average (MA) interframe predictions. MA predictive VQ executes the prediction using the code vectors (codebook outputs) in the previous frames. Moreover, this method has the following features, compared to autoregressive (AR) predictive coding, which executes the prediction based on the previous quantized (decoded) values: (1) even if a bit error is produced in the transmission channel, its effect to the succeeding frames remains finite; and (2) the stored codes can be decoded from any time point.

This paper discusses the quantization performance and the robustness against bit errors of the MA predictive VQ, as well as the training method for the codebook. Assuming an application to low‐bit‐rate speech coding, the configuration considered has a frame length of 40 ms, with each frame composed of four subframes. A method for further improvement of the efficiency is reported. As a result of evaluation using actual speech data, bit reductions of approximately 16 percent and 23 percent are achieved for the quantization for each 20 ms and 40 ms, respectively, compared to the conventional method which does not use interframe prediction.
Fuzzy programming for large‐scale multiobjective linear programming problems with block angular structure
Authors: Sakawa Masatoshi , Inuiguchi Masahiro , Sawada Kazuya
Published: July 2007

Since Dantzig and Wolfe proposed the decomposition principle for the large‐scale linear programming problem with the block angular structure in the early 1960's, there have been made a large number of intensive studies for the large‐scale mathematical programming problems with special structures. Considering the fuzziness of the human decision, however, it seems more adequate to assume that the decisionmaker has the fuzzy goals or fuzzy constraints for the objective functions and for the coupled constraints in the problem formulated as the large‐scale linear programming problem with the block angular structure.

From such a viewpoint, this paper focuses on the large‐scale multiobjective linear programming problem and proposes a new method to derive easily the satisficing solution for the decisionmaker considering his/her fuzzy goal. In the proposed method, the fuzzy goal of the decisionmaker for each objective function is specified by the linear membership function through the interaction with the decisionmaker. Fuzzy decision is applied to integrate the fuzzy goals. Then the original problem is decomposed into a master problem and several independent small‐scale linear programming subproblems.

The Dantzig‐Wolfe decomposition principle is applied, where the optimal solution as a whole is obtained through iterations, by updating the goal function of each subproblem according to the solution of the master problem. Thus, it is shown that the satisficing solution is obtained relatively easily considering the fuzzy goals of the decisionmaker.
Analysis of modified transmission line antenna for a portable telephone
Authors: Kumon Yasunori , Tsukiji Takehiko
Published: July 2007

In previous works, the authors have developed a mobile communications antenna and a high‐frequency communications antenna using the modified transmission line antenna with good results. The modified transmission line antenna is a kind of low‐profile transmission line antenna of half a wavelength. It has the advantage that desirable operating characteristics can be chosen in the antenna design because it is relatively easy to control impedance and gain by modifying its shape.

In this paper, for the purpose of developing an antenna for portable telephones, a modified transmission line antenna is proposed for a portable telephone with this antenna mounted on a conducting box representing the body of the portable telephone. This configuration was analyzed using the wire‐grid method. As a result, impedance, gain, and directivity characteristics which make this antenna suitable for portable telephones were obtained and the results are described here. Furthermore, if the conducting body representing the telephone is 100 to 160 mm long, it has a relatively small effect on the antenna characteristics, which is useful for miniaturization of the telephone body. From the calculated results and measurement results, the effectiveness of the wire grid analysis method is verified.

Finally, the case is analyzed where the modified transmission line antenna is mounted on a folding portable telephone body, which is becoming popular. It is shown that the folding angle of the telephone does not significantly affect its characteristics.
Pitch synchronous innovation code excited linear prediction (PSI‐CELP)
Authors: Miki Satoshi , Moriya Takehiro , Mano Kazunori , Ohmuro Hitoshi
Published: July 2007

This paper proposes a new speech coding method pitch synchronous innovation code excited linear predictor (PSI‐CELP). This method is based on CELP but adds pitch synchronous innovation. This results in even random codevectors being adaptively converted to have pitch periodicity for voiced frames. This scheme can improve the synthesized speech quality of voiced frames in the low bit‐rate CELP without increasing either computational complexity or bit rate.

In addition to pitch synchronous innovation, this paper also proposes four other methods in which the quality of synthesized speech is effectively improved or the computational complexity is reduced for low bit‐rate CELP: (1) the perceptual weighting filter is an MA‐type and uses nonquantized LPC parameters; (2) the fixed codebook is prepared as part of the adaptive codebook for nonperiodic frames; (3) the random codebook consists of two‐channel conjugate sub‐codebooks; and (4) codebook searches are made using delayed decision. In a speech coding system with pitch synchronization of random codevectors, improvements in ratio SNR, segmental SNR, and opinion equivalent Q value are respectively 1.6 dB, 1.0 dB, and 0.5 dB. These basic methods were adopted in the standard codec for Japanese half‐rate digital mobile telephone service.
Noise cancelling for the digital holter monitoring system
Authors: Kohama Takashi , Nakamura Shogo
Published: July 2007

The present trend is to develop a totally digital Holter electrocardiogram (ECG) recording system. To this end, research and development are in progress for the digital Holter ECG as the recorder. The conventional Holter ECG is a device which is carried for long hours and records the ECG waveforms on magnetic tape as the analog signal. It has problems in accuracy, e.g., waveform distortion, moving parts such as the motor, as well as volume and weight.

The digital Holter ECG considered in this paper is the simplified digital Holter ECG which converts ECG waveforms for 24 hours into digital form, to record and process various waveforms. By totally designing the system in a digital form, the circuit can easily be integrated into LSI. By using the IC memory card as the recording medium, a small‐sized and lightweight equipment can be realized. Thus, the problems in the past system can be solved.

A problem in the present IC memory card is that the recording capacity is limited; and the data compression of approximately 1/60 is required to record ECG waveforms for 24 hours. The achievable compression ratio is affected by the noise on the ECG waveform and the fluctuation of the baseline.

This study is concerned with the method to eliminate the noise, which is a problem in the data compression. The results of analysis and simulation are presented.
Generalization of smoothness measure for standard regularization and its application to surface reconstruction
Authors: Mizutani Hiroyuki
Published: July 2007

The standard regularization is a theory that converts ill‐posed problems into well‐posed problems by assuming some a priori information about solutions. Usually, this assumption is concerned with the smoothness of a function that is a solution of the problem. To measure the smoothness of a function, a membrane or thin‐plate measures have been used, i.e., the deformation energy of a membrane and a thin‐plate, respectively. The smoothness measure is generalized by using a continuous order differential‐integral operator. This generalized measure is applied to surface reconstruction from sparse data and its usefulness and superiority to these measures are shown.
Problem of computing the expected maximum number of vertex‐disjoint s‐t paths on probabilistic digraphs
Authors: Cheng Peng , Masuyama Shigeru
Published: July 2007

The problem of finding vertex disjoint paths (i.e., paths sharing no vertices) between a specified pair of vertices of a digraph has been extensively studied as one of the fundamental problems in graph theory. For digraphs whose vertices or edges have mutually independent failure probabilities (called probabilistic digraphs hereafter), the expected maximum number of vertex disjoint paths is a measure of reliability of the network and its computation is an important practical problem. In this paper, the problem of computing the expected maximum number of vertex disjoint paths between a specified pair of vertices in a given probabilistic digraph is considered. It is shown that the problem is in general NP‐hard but that the expected value can be obtained in polynomial time for special types of probabilistic digraphs such as probabilistic s‐t series‐parallel digraphs and probabilistic s‐t complete multistage digraphs in which the vertices have the same failure probability and the arcs are failure‐free. Further, the problem is shown to be NP‐hard for the following three cases: probabilistic planar digraphs; probabilistic s‐t out‐in bitrees; and probabilistic s‐t complete multistage digraphs.
The analysis of nonstationary processes based on a kinetic model
Authors: Takizawa Yumi , Fukasawa Atushi , Oda Keisuke , Harashima Hiroshi
Published: July 2007

This paper presents a new analysis method for the nonstationary phenomenon using modeling. Method efficacy is investigated and effectiveness is demonstrated. Nonstationarity is modeled into the kinetics of the generating process. the behavior of the nonstationary process is represented by the characteristics at a point on the time axis (instantaneous characteristic) and the characteristics of its time variation (time‐variant characteristics). By estimating the nonstationary spectrum, the instantaneous characteristics are determined and then the time‐variant characteristics are analyzed by applying the kinetic model. the nonstationary phenomenon is first modeled using kinetic energy. Characteristics at a point on the time axis are regarded as a point of mass. Position, velocity, and acceleration are determined; these parameters signify a model with time‐varying characteristics. the arising time of the event is determined, based on the relations among the parameters. the characteristic plane is constructed using these parameters, and nonstationarity is evaluated on the plane. the proposed algorithm is then evaluated using an artificial signal and the model is validated using a speech signal. This analysis yields a new finding that deals with the characteristics of the phoneme, thus demonstrating the usefulness of the proposed method.
Hopfield‐type networks: Making the condition of learning pattern more lenient by threshold
Authors: Aizawa Tadashi , Hyogo Akira , Sekine Keitaro
Published: July 2007

The information storage algorithm of a Hopfield‐type network is a kind of self‐correlation learning. This algorithm needs the condition in which vectors of learning patterns cross at right angles to each other. This paper proposes a method to make this condition easier. To this purpose, threshold is used. An investigation of how to decide quantity is included. This paper discusses a network with units whose state is 1 or 0. Fixing method of threshold has not been proposed before to this type of network. It is, therefore, proposed in this research; the goal is to facilitate learning patterns.
Decomposition of flowcharts using a digraph hierarchical property
Authors: Kato June , Miyake Nobuhisa
Published: July 2007

This paper discusses the hierarchical properties of digraphs and proposes flowchart decomposition algorithms using these properties. the hierarchical structure discussed here is based on only the topological relationships between vertices and the inclusive relationship among sets of vertices. When the given flowchart represents the control flow of a computer program, this hierarchical structure also shows strong correspondence between these inclusive relationships and program modules. the proposed flowchart decomposition algorithm consists of one algorithm for detecting modules and another algorithm for further decomposing the detected modules. This paper focuses on the former algorithm.

Because the hierarchical structure discussed in this paper corresponds to the traditional program modules (e.g., function or procedure) of most of the computer programming languages, the module‐detection algorithm can be used not only for decomposing flowchart but for more general purposes, including module design. Two types of module‐detection algorithm are discussed in this paper. One detects all modules, and the other detects a restricted subset of modules from the standpoint of flowchart decomposition. We then demonstrate with a practical example that the latter algorithm is ten times faster than the former.
An effective method to find all solutions of piecewise‐linear circuits by using partitioning
Authors: Mizutani Hikaru
Published: July 2007

With the expansion of the circuit scale in recent years, the dimension of the circuit equation to be analyzed also is increased, which requires a tremendous amount of time in the processing. In addition, the circuit often contains a large number of nonlinear elements, which may result in nonunique multiple solutions.

This paper aims at the solution of such a problem and proposes an algorithm as a means to determine all solutions of the nonlinear circuit with the piecewise‐linear approximation with a high speed. In the proposed algorithm, the circuit is partitioned and each subnetwork is analyzed. In the proposed analysis, the relation is derived for each of the linear regions of the subnetwork, between the current flowing from other subnetworks at the partitioning points and the node voltages.

The existence of the solution is examined for each linear region. Then the subcircuits are connected successively and the current‐voltage relation is derived for a larger subnetwork. Only the linear regions which have the possibility of containing a solution is considered in the calculation, and it is expected that the computation will be completed at a high speed.

By iterating the connection process, all solutions for the circuit can eventually be obtained. Since the subcircuit contains fewer nodes and fewer linear regions, it is expected that the analysis of the subcircuit can be executed at a high speed.
Image coding and decoding by discrete‐time cellular neural networks
Authors: Ikegami Munemitsu , Tanaka Mamoru
Published: July 2007

When the original signal to be encoded has a strong correlation in the neighborhood, as in the case of the natural image, it can be expected that the original signal is represented by a smaller number of bits.

This paper proposes the following low‐bit coding system. the analog image is converted into the digital image by A‐D conversion in the transmitter based on the dynamics of the locally interconnected discrete‐time cellular neural network (DTCNN). the digital image is passed through a lowpass filter in the receiver so that the original analog image is restored. In this method, the coding of the static image is reduced to the optimization problem where the error between the original image and the decoded image is minimized. the distortion function is defined so that the noise generated by encoding is suppressed from the viewpoint of the whole image.

A high‐quality image is reconstructed by the dynamics of the multivalued neuron, which is an extension of the output of the neuron to multiple values. In this method, a parallel processing dynamics is employed, where a complex function can be realized from simple devices, and it is expected to realize the digital conversion in real‐time of the analog information which is input in parallel, as in the case of the light input to the retina.
An extended overlap‐add method and overlap‐save method for sampling rate conversion
Authors: Muramatsu Shogo , Kiya Hitoshi
Published: July 2007

The overlap‐add (OLA) and overlap‐save (OLS) methods are well known as efficient schemes for higher‐order filtering. In those methods, the linear convolution for the infinite‐duration data is reduced to the circular convolution for the finite blocks, and the fast‐Fourier transform (FFT) is applied to the execution of the circular convolution.

This paper proposes new OLA and OLS methods, which can directly be applied without a redundancy to the rate conversion. First, the conventional theory is extended for the finite‐duration data. Then, the result is generalized for the infinite‐duration data. Two methods are proposed as the main result in this paper, which are the extended overlap‐add and extended overlap‐save methods. Finally, the proposed method is compared to the polyphase structure, which is well known as an effective realization method for the rate conversion.
Noise reduction in state‐space digital filters using high‐order error feedback
Authors: Hinamoto Takao , Karino Shuji
Published: July 2007

A technique for the reduction of the output noise variance in state‐space digital filters is developed using a high‐order error feedback consisting of a finite impulse response (FIR) digital filter. First, the optimal error feedback structure using all of the elements of the coefficient matrices in the FIR digital filter is derived. Next, the suboptimal error feedback structures are shown; the elements of the coefficient matrices in the FIR digital filter are constrained to be either symmetric or antisymmetric. Finally, a numerical example is given to illustrate the utility of the proposed technique.
Bounded real property of time‐delay time‐varying systems and its application to learning control
Authors: Yashiki Satoshi , Matsumoto Naoki , Endo Tetsuro , Takahashi Tohru
Published: July 2007

Learning control is one of the effective control methods to achieve high‐speed motion and accurate positioning of mechanical systems which repeat the same motion. In this method, the control input is improved in each trial based on the error between the desired trajectory and the system output. the learning control system is stable if the error transition system defined in learning control is bounded real (BR).

Kawamura et al. have shown the stability condition of the learning control for time‐invariant continuous‐time systems based on BR property. Takahashi has studied the stability of learning control in the case of time‐varying continuous‐time systems and time‐varying discrete‐time systems.

This paper derives the stability condition of the learning control for time‐delay time‐varying (TDTV) systems which have not been reported so far because of its complexity. First, the sufficient condition for TDTV systems to be BR independent of delay is derived and the circuit's theoretical interpretation for the BR condition is given. Using this result, the stability condition for learning control for TDTV systems is given. the error of learning control converges to zero by increasing the number of the trials if the corresponding error transition system, which is defined from the error at thei‐th trial and error ati+ 1‐th trial in the learning control, is BR. Finally, a numerical example of the learning control for the TDTV system which satisfies the obtained stability condition is demonstrated, showing that the derived stability condition is valid.
A method of designing filter banks with alias‐free characteristics at equally spaced frequency points
Authors: Kiya Hitoshi , Yamazaki Hirotatsu , Yamada Yoji
Published: July 2007

It is known that the quadrature mirror filter (QMF) bank used for subband coding cannot avoid the aliasing effect for adaptive signal processing or spectral estimation, etc. A new class of filter banks is desired for these applications where the effect of aliasing should also be considered.

In this paper, a new class of filter bank is reported which has alias‐free characteristics at equally spaced frequency points. This filter bank is called frequency sampling filter bank, and it consists of nonrecursive FIR filters. This paper also reports design methods for two types of the filter banks, keeping in mind the reduction in amount of calculation for practical implementation. In one method, all channel filters of the filter bank are determined by complex exponential modulation of a prototype filter. the other method determines the channel filters by cosine or sine modulation of a prototype filter. Both designs have the forementioned aliasing characteristics, and an efficient structure based on polyphase configuration is possible.
Analysis of the equilibrium states of the hopfield associative memory using the system of walsh functions
Authors: Aoki Hiroyuki
Published: July 2007

This paper considers the situation where the multiple state‐patterns satisfying the orthogonality condition are memorized in the Hopfield associative memory model. Also discussed is the relation between the equilibrium state produced in the associative memory model and the memorized patterns. Numerous studies have been made of the associative memory model by the statistical approach. By contrast, this study considers the problem from the algebraic standpoint, using the Walsh functional system.

As the first step, a sufficient condition is derived for the associative memory model to be in the equilibrium state. Using this condition, the equilibrium states are classified into three types and the relation between the memorized pattern and the generated equilibrium state is analyzed. As applications, a method of estimating the number of equilibrium states and their state pattern from the data concerning the memorized pattern, as well as a method of making the given pattern an equilibrium state, are shown through examples. the results are obtained by the component analysis of the state of the associative memory model using the Walsh functional system. In other words, the usefulness of applying the Walsh functional system is demonstrated.
Image restoration using layered neural networks and hopfield networks
Authors: Muneyasu Mitsuji , Yamamoto Kazunari , Hinamoto Takao
Published: July 2007

An algorithm is developed for the restoration of an image degraded by a known two‐dimensional (2‐D) shift‐invariant point‐spread function and corrupted with white noise. A layered neural network and the Hopfield network are used for the edge detection, and the restoration and smoothing of a blurred image, respectively. In particular, a layered neural network is proposed for exact edge detection where the inputs consist of three pixel values and a local variance in a 2‐D mask. This network can detect edges and suppress the noise in an image at the same time, and its performance is adjusted by learning. Finally, some examples are given to illustrate the utility of the proposed algorithm.
Measurement and analsis methods for critical flicker frequency and observer fatigue caused by television watching
Authors: Mitsuhashi Tetsuo
Published: July 2007

Aiming at the quantitative measurement of the fatigue of the television observer, a new practical and simple method of critical flicker frequency (CFF) measurement and analysis is considered. As to the measurement, a method is considered where the CFF at the start of the measurement is used as the initial value and the relative change is measured in a short time with a high accuracy.

It is verified experimentally that the measurement by the descending series of the method of limit is suitable. As to the analysis, two methods are compared. One is to normalize the average of measured values of all the observers, and the other is to normalize the measured value of each observer and then to calculate the average. It is shown as a result that the latter can represent more clearly the change of CFF. Then the effectiveness of the proposed method in the measurement of the fatigue of the television (TV) observer is examined. CFF of the actual TV image observer is measured, including the binocular three‐dimensional (3‐D) TV. As a result, it is verified statistically that CFF depends on the observation time and the image display method. the result corresponds well to the subjective evaluation of the fatigue. Thus, it is concluded that CFF can be a useful measure in assessing quantitatively the fatigue of the TV image observer, including the binocular 3‐D TV. It is expected that the image quality and the improvement effect of the observation condition can be examined quantitatively.
Linear‐phase biorthogonal wavelet function with arbitrary regularity
Authors: Yasuoka Hiromichi , Sakamoto Yuuji , Ikehara Masaaki
Published: July 2007

This paper proposes a method of derivation for the linear‐phase biorthogonal wavelet functions with an arbitrary regularity, based on the finite impulse response (FIR) half‐band digital filter. Within the range of the orthogonal wavelet functions at present, the Haar function is the only one that satisfies the linear‐phase property. In the image processing and other applications, the linear‐phase property is more important than the orthogonality.

From such a viewpoint, the biorthogonal wavelet functions are proposed, for which the orthogonality condition can be relaxed and the linear‐phase property can be emphasized. In its derivation, however, mathematical sophistication is involved and the range of the derived function is considerably limited. With the development of the wavelet transform in recent years, it has become more important to select the base function flexibly according to the purpose of the processing and it is desired to establish a derivation method for the more general biorthogonal wavelet function with a higher degree of freedom.

This paper presents a derivation method for the more general biorthogonal wavelet function with an arbitrary regularity. the method is based on the half‐band digital filter, and the flat response is added to the filter composing the wavelet function. Then the regularity condition for the function is considered, and the amplitude square‐error is used as the evaluation function. Finally, several design examples are shown for the linear‐phase wavelet function derived by the proposed method.
Classification of ultrasonic images using fuzzy reasoning and spatial smoothing effect of textural features
Authors: Mochizuki Takashi , Ito Masayasu
Published: July 2007

This paper discusses the method for the semiautomatic discrimination of the biological tissue and organs, based on the medial ultrasonic data. the ultrasonic image is difficult to be evaluated on the absolute basis, since multiple physical information is included simultaneously and the image is affected greatly by the individuality of the object. This method utilizes the feature parameters of the image for various tissues, and the discrimination is executed on the relative basis by comparing the feature parameters.

In the past method, the general procedure is to utilize the first‐order statistics such as the average image density. In the case of a tumor in the liver, for example, there does not exist a remarkable density difference between the image of the tumor and the surrounding normal tissues, which prevents a satisfactory discrimination of the tumor and other images.

This paper aims at the discrimination of tissue images with small density differences, based on the textural feature parameters (the second‐order statistics) contained in the image. However, the crisp‐like discrimination is difficult since there is an overlapping ambiguous area between the histograms of the feature parameters of the tissues. to improve the discrimination rate for the ultrasonic image, a method is proposed which utilizes the nonlinear input‐output relation of the fuzzy inference and the spatial distribution of the image to be adapted to each object. Using the data obtained from the test phantom for the ultrasonic diagnostic equipment and the biological data obtained from four subjects, the effectiveness of the proposed method is demonstrated.
Improved methods for deriving the complete weight distribution of reed‐solomon codes
Authors: Tokiwa Kin‐Ichiroh , Tanaka Hatsukazu
Published: July 2007

The authors previously considered the problem of deriving the complete weight distribution (CWD) of Reed‐Solomon (RS) codes and developed a formula which established the relationship between a specific codeword set and the CWD of an RS code. If this formula is used, then the CWD of an RS code can be determined by examining fewer codewords than would be required by an exhaustive search. However, subsequent investigation showed that this formula can be applied in its current form only to codes with prime length and not to codes with composite length. A new form of this formula given herein does not depend on whether the code length is prime or composite. the set of codewords that must be examined to determine the CWD of an RS code is respecified and a formula that gives the relationship between this newly specified codeword set and the CWD of an RS code is derived. It is also shown that, when the generator polynomial of an RS code does not have unity as a root, then the number of codewords that must be examined can be reduced even further.
A fuzzy dual decomposition method for large‐scale multiobjective nonlinear programming problems with block angular structure
Authors: Sakawa Masatoshi , Yano Hitoshi , Sawada Kazuya
Published: July 2007

This paper considers the large‐scale multiobjective nonlinear programming problem with block angular structure. A new method is proposed for deriving the compromising solution of the decisionmaker based on the fuzzy decision, considering the fuzzy goal of the decisionmaker for each objective function. the large‐scale nonlinear programming problem (original problem) for deriving the solution based on the fuzzy decision is formulated. Then, based on the dual decomposition technique introduced by Lasdon, the dual problem for the original problem is formulated.

A two‐level optimization algorithm is proposed where the formulated dual problem is decomposed and the solution for the original problem is obtained by iteratively solving the subproblems. the validity of the algorithm is examined through a simple numerical example with the block angular structure.
Edge detection using markov random field models—optimization and parameter estimation by mean field annealing
Authors: Kudo Hiroyuki , Saito Tsuneo , Kawauchi Michiko
Published: July 2007

An edge‐detection method based on Markov random field and statistical estimation is proposed. Compound Gauss‐Markov random fields are used as the image model, and the edge‐detection problem was formulated as the maximuma posterioriprobability (MAP) estimation problem which can determine the estimated value of line process from the observed value of intensity process. the simulated annealing (SA) method and iterative conditional model (ICM) method are well known as relaxation methods for such types of problems. However, a large amount of calculations is needed in the SA method and convergence to local optimum solution is a major problem in the ICM method.

This paper determines the MAP estimation solution using the method of mean field annealing (MFA), which has shown good results for the combinatorial optimization problem. Moreover, since in this method the shape and continuity of the detected edge changes with the model parameters of compound Gauss‐Markov random field, parameter estimation is an important problem.

In the previous research, those parameter estimation methods have been studied where parameters determined from experience have been used or parameters are estimated using the edges traced by a human being. This paper proposes the method of automatic parameter estimation from the observed image based on the maximum likelihood (ML) criterion. Moreover, the gradient descent method and expectation maximization (EM) algorithm are used to calculate the actual ML solution.
High‐frequency compensation of frequency‐dependent negative resistance
Authors: Taki Masahiro , Hayahara Etsuro
Published: July 2007

The GIC filter is one of the LC simulation active filters and has the advantages that the element sensitivity is low and the higher‐order filter can easily be constructed compared to the single‐amplifier active filters. For the high‐frequency filter, however, the filter performance may be deteriorated due to the frequency dependency of the operational amplifier represented by the GB product, for example. Several compensation methods have been proposed for the single amplifier filter, while no effective method has been proposed for the GIC.

From such a viewpoint, this paper proposes a compensation method for the frequency response of FDNR composed of GIC. the proposed frequency‐dependent negative resistance (FDNR) is applied to the actual filter circuit, and it is confirmed that the circuit operates almost as is predicted by the theory.
Performance of an adaptive viterbi decoder employing respective‐state channel estimation
Authors: Kubo Hiroshi , Murakami Keishi , Fujino Tadashi
Published: July 2007

This paper discusses a performance of an adaptive Viterbi decoder, which performs the carrier recovery and the Viterbi decoding simultaneously. the authors have already considered the combined maximum‐likelihood sequence estimator (MLSE), according to trellis combined equalization and the Viterbi decoding and showed that the bit‐error rate in the fading environment can be improved by applying channel estimation respecitvely for states of the Viterbi algorithm. the adaptive Viterbi decodere discussed in this paper corresponds to the combined MLSE in the case of no intersymbol interference and and easily be extended to the case of the diversity reception.

First, the square Euclidean distance is derived approximately by combining the channel estimation by the least mean squares (LMS) and MLSE, and it is shown that the adaptive Viterbi decoder can perform the diversity combining without carrier phase information, with only a little performance degradation compared to the ideal Viterbi decoding with known carrier phase. Then, by a computer simulation, the results of analysis and the numerical calculation are confirmed and the proposed adaptive Viterbi decoder is shown to be effective in the carrier frequency offset and fading environment.
Error spectrum shaping in two‐dimensional digital filters
Authors: Hinamoto Takao , Karino Shuji , Kuroda Naoki
Published: July 2007

A technique for minimizing the normalized noise variance at the output of two‐dimensional (2‐D) recursive digital filters is developed where error spectrum shaping of the filter is performed by means of higher‐order feed‐back. First, the optimal solution of error feedback coefficients is derived. Then suboptimal error feedback coefficients are found in which the error feedback coefficients are symmetric. Here, both the optimal and suboptimal solutions are found analytically by solving the minimization problem of a quadratic form. Finally, a numerical example is given to check the validity of the proposed algorithm and it is shown that the proposed technique is effective even if the optimal or suboptimal coefficients are rounded to a power of 2.
A new normalized stochastic approximation algorithm using a time‐shift parameter
Authors: Okamura Atsushi , Kirimoto Tetsuo , Kondo Michimasa
Published: July 2007

This paper proposes a stochastic approximation adaptive algorithm aiming at a fast convergence speed, where the step size is normalized by the instantaneous power ‖x(k)‖2 of the input signal and a constant timeshift β is included at the weight update, as α/(k + β)‖x(k)‖2. It is shown first that the proposed algorithm gives a convergence to the Wiener solution. Then the transient response of the mean error is analyzed. It is shown as a result that β ⩾ α/2 is the condition for the time‐shift parameter to guarantee the uniform convergence of the algorithm.

It is shown also that the speed of convergence is improved greatly compared to the conventional algorithm when α is set large enough so that the product of α and the minimum eigenvalue of the autocorrelation matrix exceeds 100 and the time‐shift parameter is set as β ∼ α/ 2. the performance is evaluated by a computer simulation. the above result of analysis is verified, and it is shown that the proposed algorithm can realize the same convergence speed as that of the recursive least‐squares (RLS) algorithm.
Computational complexity of multicommodity flow problems with uniform assignment of flow
Authors: Ito Hiro
Published: July 2007

This paper treats multicommodity flow problems with a flow assignment constraint. General multicommodity flow problems require both the path set in which the commodities are assigned and the volume to which the commodities are assigned for each path. However, in this paper, we treat only problems wherein paths are required. the flow is assigned to the paths uniformly. This problem appears in state‐and‐time‐dependent routing (STR), which is a kind of dynamic routing scheme in telecommunication networks. We show that this problem is NP‐complete and remains NP‐complete if the lengths (number of edges) of paths are restricted to less than or equal to two but can be solved in polynomial time if the path lengths are restricted to be less than or equal to two and the number of commodities is fixed.
Accurate measurement of vocal tract shapes using magnetic resonance imaging
Authors: Yang Chang‐Sheng , Kasuya Hideki , Kano Sigeru , Sato Toshihiko
Published: July 2007

In this paper, a method is developed for an accurate measurement of the three‐dimensional shape of the vocal tract, using magnetic resonance imaging (MRI), aiming at the analysis of the features of the vocal tract shape in the utterance by children, males and females. to derive the precise data for the shape of the vocal tract, the data for the vocal tract of the oral cavity is derived from the coronal MR image from the lips to the atlas. the data for the vocal tract from the glottis to the palate are derived from two kinds of horizontal MR images. High‐speed photography is used, and the imaging time is 2.9 s for each slice. the central line of the vocal tract is estimated from the mid‐sagittal MR image, and the vocal tract length is calculated. Based on the mid‐sagittal MR image, the three‐dimensional shape of the vocal tract is reconstructed from the vocal tract contour in the three kinds of images.

To process the large amount of data, a method is developed that automatically extracts the vocal tract contour from the MR image. A correction method is proposed for the teeth shape which cannot be imaged by MRI, based on the MR image of the dental impression. the error of the measuring method is evaluated quantitatively using a model for the vocal tract. the transfer function for Japanese /a/ are estimated by the proposed measuring method, and the result is compared to the result of LPC analysis of the speech recorded at the site. It is seen as a result that the errors for F1, F2, and F3 are 8, 17, and 141 Hz, respectively.
Two decodable code pairs for the two‐user binary adder channel
Authors: Shi Jian‐Jun , Watanabe Yoichiro
Published: July 2007

This paper discusses the construction method for the two‐decodable code pair (C, S) for the two‐user binary additive channel as a model for the multipleaccess channel. (C, S) is constructed by defining the second code S as the independent set of the two‐order associated graph of the first code C. By restricting the code C to a particular linear code, it is possible to give an evaluation of the lower bound for the independence number of the two‐order associated graph. In several construction examples for the code pairs, the derived lower bound coincides with the independence number.
A design technique for linear‐phase finite impulse response digital filters with constant passband amplitude
Authors: Tsuchiya Mamoru
Published: July 2007

This paper describes a new design algorithm for finite impulse response (FIR) digital filters, which approximates constant passband amplitude and linear phase characteristics. The algorithm converts a filter's specification characteristic into a continuous impulse response by using the frequency response of a window function. The continuous impulse response is translated into discrete values to calculate tap‐gain coefficients of an FIR filter. In this design algorithm, window functions belonging to the Hanning window series (such as the Harris' window whose characteristics are already known) are used. This design algorithm is an analytical design technique, featuring a filter length that can be decided when the specification is decided and does not require iterative numerical calculations to calculate tap‐gain coefficients.