bluuebunny/crossref_metadata_2025 huggingface dataset:
Behavioral verification of cpus using functional information extraction
Authors: Ohmura Masahiko , Tamaru Keikichi , Yasuura Hiroto
Published: July 2007

The information extraction techniques which transform the description at the logic circuit level in the LSI design process into the description at the functional level are important. The authors have been studying the automatic transformation technique. By combining that technique with the technique for automatic logic synthesis which is now being developed, it is expected that the descriptions at the logic circuit level and the functional description can be transformed to each other. This will help to construct the integrated CAD system.

The feature of the method proposed in this paper is that the arithmetic operation functions and the functions at the RT level can be extracted from the tremendous amount of complex circuit descriptions and the results are represented in a simple and clear form. Consequently, the method is especially suited to the behavior verification of the circuits, such as CPUs.

In this paper, the object of consideration is limited to the circuits in the CPU system and the application of the functional information extraction technique to the design verification is discussed. Since the behavior description for a CPU is given in a simple and clear form, using the instruction code table or state transition table, the design verification is easy by a visual comparison of the extracted functional description.

The complex logic functions in the control circuit are compared using BDD. An actual example of verification is shown in this paper for the 8‐bit microprocessor, and it is demonstrated that a practical processing is realized from the viewpoints of required time and memory capacity.
Application of analogical reasoning and extraction of tutoring rules for concept‐formation learning
Authors: Okamoto Toshio , Morihiro Koichiro , Matsuda Noboru , Takuma Shimpei
Published: July 2007

The concept formation in humans is made through a process where the common items and rules in the observed events are extracted. The observed events may contain an event which is not consistent with the already formed concept. In such a case, the modification or restructuring of the concept takes place and an effort is made toward differentiation or recombination of the concept.

With respect to the process where a human forms or acquires a concept, Klausmeier et al. considered the development of the levels in the concept formation. They defined four levels which are the real level, equivalence level, analogy level, and formal level and discussed the order among the levels. In this study, the inference engine is built into the expert module of ITS and the system model to support the concept formation is considered, which contains the intellectual function corresponding to the analogy level.

As a domain to handle a realistic problem, a simple transformation of a geometrical figure is considered and the following functions are realized: (1) construction of analogical reasoning engine in the transformation of geometrical figures, (2) construction of the student model which represents the student's analogical reasoning process, and (3) generation of problems based on the student model.
Synthesis of separable‐denominator multidimensional digital lattice filters
Authors: Watanabe Eiji , Nishihara Akinori
Published: July 2007

Digital filters play an important role in multidimensional signal processing. Because the coefficients must be expressed in finite wordlength, the deterioration of the characteristics becomes a problem. Using a lattice structure whose efficiency has been confirmed for one‐dimensional (1‐D) digital filters, this paper proposes a unified synthesis method for the practically important separable‐denominator multidimensional IIR digital filters.

First, the implementation of a 2‐D separable‐denominator transfer function by a separable‐denominator 2‐D digital lattice filter is considered. A characteristic of the proposed separable‐denominator 2‐D digital lattice filter is that both the usual tapped lattice structure and the injection‐type lattice structure are used and the signal flow becomes simple compared to the method so far.

Next, when the numerator polynomial is a mirror‐image polynomial, an implementation method is presented in which the number of tap multipliers is reduced significantly. Furthermore, a method is suggested for the synthesis of a 3‐D transfer function based on the same approach as for the 2‐D case. Finally, the efficiency of this method is confirmed by an example.
Fundamental study of periodically time‐varying state‐space digital filters—statistical analysis, scaling, and stability
Authors: Kawamata Masayuki , Higuchi Tatsuo , Yang Xuedong
Published: July 2007

Analyses of periodically time‐varying (PTV) digital filters that have been published deal with the fundamental properties and the design methods of PTV digital filters described by transfer functions. However, few studies have been published on the design of high‐precision and high‐efficiency PTV digital filters with minimum roundoff noise, minimum coefficient sensitivity, and model reduction.

To analyze these design problems, this paper reports on the statistical analysis, scaling, and stability of PTV state‐space digital filters. First, PTV state‐space digital filters are described and dual PTV state‐space digital filters and equivalent transformations of PTV state‐space digital filters are defined. Then covariance of the states and variance of the output in response to a white Gaussian noise input are analyzed statistically. Based on these results, a scaling analysis is presented. Also, asymptotic stability and bounded input‐bounded output stability of PTV state‐space digital filters are discussed. Finally, a numerical example is given to verify the foregoing analytic results.
Control of second‐order temperature coefficients of delay time for saw propagating in thin‐film layers
Authors: Nakagawa Yasuhiko , Yoneda Toshimaro , Omata Tadahiro , Hosaka Keiko
Published: July 2007

In a surface acoustic wave substrate with a “piezoelectric thin‐film/substrate” configuration, the temperature coefficients of the acoustic constants of the Ta2O5 and ZnO thin films were determined by means of the least square method based on the experimental values of the temperature coefficients of the acoustic constants. Also, with regard to the two‐layered model of a “piezoelectric thin‐film/piezoelectric thin‐film/substrate” configuration, the surface acoustic wave speed, electromechanical coupling coefficient and temperature characteristics were computed and the combinations of the film thicknesses were derived for a first‐order zero‐temperature coefficient. Further, it is recognized that the ZnO thin film and the Ta2O5 thin film have second‐order delay time temperature dependences with opposite signs and then a method is proposed for controlling the first‐ and second‐order temperature coefficients in a two‐layered substrate. It is found experimentally and theoretically that there exists a highly stable substrate which exhibits a third‐order characteristic at an appropriate film thickness.
m‐route flows in a network
Authors: Kishimoto Wataru , Takeuchi Masashi
Published: July 2007

The m‐route flow between two vertices is defined in a flow network which is an extension of the flow between two vertices. In the communication network, the m‐route flow corresponds to the set of m‐route structured communication paths. In a flow network, the set of m‐edge disjoint paths connecting the two vertices considered and composed of edges with the same capacity is defined as the elementary edge‐m‐route flows between those two vertices.

For the cut of the flow network, the edge‐m‐route capacity is defined as an extension of the ordinary capacity and it is shown that a theorem similar to the max‐flow min‐cut theorem as in the ordinary theory of flows applies to the maximum edge‐m‐route flow. It is shown also that, when the set of m internally disjoint paths connecting the two vertices considered and which is composed of edges of the same capacity is defined as the elementary vertex‐m‐route flow, the vertex‐m‐route flow between two vertices is defined as the sum of elementary vertex‐m‐route flows between those two vertices; and a similar theorem holds. In the communication network, the maximum m‐route flow corresponds to the maximum number of communication paths that can be composed of the m‐route structure between the two terminals.
A consideration for deriving nonbinary irreducible polynomials
Authors: Sugimura Tatsuo , Suetugu Yasunori
Published: July 2007

An irreducible polynomial can be derived in a stochastic way by examining the irreducibility of randomly generated polynomials. On the other hand, a systematic method of derivation for the irreducible polynomial has recently been introduced by presenting a class of irreducible polynomials of particular forms.

This paper shows clearly that a higher‐order irreducible polynomial can be derived by applying a suitable variable transformation to the given irreducible polynomial. It is shown also that, when an irreducible polynomial is given where the coefficient is the element of a finite field with odd prime characteristic, an infinite number of higher‐order irreducible polynomials can be derived from that polynomial. A precise algorithm for the derivation is shown.
Synthesis of variable IIR digital filters with complex coefficients
Authors: Murakoshi Nobuo , Nishihara Akinori , Watanabe Eiji
Published: July 2007

Sometimes it is required to change the filter characteristics of a digital filter in real time during its filtering operation. To meet this requirement, variable cutoff frequency lowpass filters with real coefficients have been reported using only one parameter. Moreover, variable bandpass filters with complex coefficients have been reported where only one parameter was to be changed and where either the upper or lower band‐edge frequency of the passband was fixed. However, in those filters, the frequency response was degraded with the change in characteristics which could not be resolved.

This paper proposes the synthesis of a variable IIR digital filter using one parameter and the frequency response is not degraded when the characteristics are changed. In the variable filter proposed here, the cut‐off frequency and the bandwidth may be changed without any restriction. Moreover, a simple circuit realization technique is proposed so that the filter circuit may actually be implemented. In this technique, any type of real coefficient filter circuit may be used as the prototype filter. Finally, the simulation of the synthesis problem is carried out and the validity of the proposed method is confirmed.
Verification of the possibility of separation of sound source direction via a pair of pressure microphones
Authors: Koike Norio , Yanagisawa Takesaburo
Published: July 2007

In this paper, a microphone system with which the directional separation angle can be set variably is proposed with a pair of sound pressure microphones necessary as a minimum to separate and detect the sound source. After the operating principle is described, an experimental verification is carried out. It is confirmed that the separation and detection of the sound sources are possible by the present system for the sound fields constructed by several sound sources with spectral components different from the single source.
Performance evaluation and improvements of ARQ schemes for optical communication
Authors: Uyematsu Tomohiko , Mitsugi Jun , Sakaniwa Kohichi
Published: July 2007

The optical communication system can provide highly reliable and large‐capacity transmission. It is becoming important as the major technology of highspeed digital communication. In the optical communication system, even though it is highly reliable, inevitably the bit error rate is deteriorated with the increase of the transmission rate or the repeater distance. It is anticipated that some methods of error control will be required in the future.

This paper discusses the application of the automated repeat request (ARQ) scheme, which is one of the error control methods, to the optical communication. It is noted first that the error that occurs in the optical communication channel is the transition from symbol 1 to “0,” but it can never be the transition from “0” to “ 1.” An ARQ scheme is proposed using the m‐out‐of‐n code which can detect any error in such a channel.

A performance evaluation is presented based on the throughput and the information efficiency per photon. It is shown as a result that the SR‐M3BN scheme is the best among various ARQ schemes from the practical viewpoint. It combines selective‐repeat (SR) and go‐back‐N (GBN). Then a new error‐detecting scheme is proposed based on the characteristics of the optical channel, which can reduce the number of repeats of the codeword. It is shown that, by combining the proposed scheme with SR+GBN, a higher throughput can be achieved.
Steady‐state analysis of closed‐loop switching circuit by calculating convolution integral in the steady state
Authors: Suetsugu Tadashi , Mori Shinsaku
Published: July 2007

This paper discusses the convolution integral of the circuit variable and the transfer function in the steady state of a circuit. It is shown that the integral can easily be calculated by the Shooting method when the transfer function to be calculated has a simple form. Using the proposed method of calculation, the steady‐state operation is analyzed for the periodic switching circuit which is controlled by the closed loop, such as the switching converter. A method is proposed that precisely determines the steady‐state operation of the overall circuit, while precisely analyzing the steady‐state operation of the loop circuit.

The proposed method differs from the conventional method in that the circuit, including the loop circuit not analyzed but the loop‐control circuit is represented by a transfer function separately from the overall circuit and the steady‐state response is determined by the convolu‐tional integral. Consequently, for closed‐loop switching circuits of various structures, the operation can be determined precisely without enlarging the circuit scale to be analyzed. Another point compared to the past method is that the operation specifications for the circuit can be handled with a better transparency.
Prosodic phrase segmentation based on pitch‐pattern clustering
Authors: Nakai Mitsuru , Shimodaira Hiroshi , Sagayama Shigeki
Published: July 2007

This paper proposes a method to detect the pros‐odic phrase for speaker‐independent continuous speech. The method uses continuous adjustment of a pitch pattern with the learning accent pattern (pitch template). The accent phrase with one accent cue and its corresponding pitch pattern are considered correlated. Distinctive pitch templates are obtained by classifying the pitch patterns corresponding to the accent phrase with the help of clustering.

Since pitch patterns of continuous speech are connected to these pitch templates, prosodic phrase segmentation can easily be done by One‐Stage DP matching of the pitch pattern with pitch templates. The method can also be applied to unspecified speakers whose average pitch frequency differs from each other by sliding the pitch template along the vertical axis during the matching.

The ATR continuous speech database (10 speakers, 503 sentences) is used for the experiments. For eighc templates, about 65 percent of the automatic prosodic phrase segmentations were correct and the experiment was successful in detecting about 83 percent of the prosodic phrase boundaries.
Error estimation for IDCT finite word‐length calculation
Authors: Mochizuki Takashi
Published: July 2007

Calculation error estimation formulas for limited‐accuracy operations of inverse discrete cosine transform (IDCT) are derived, considering the nonlinear property of integer rounding. CCITT Recommendation H.261 specifies a calculation error evaluation method for IDCT schemes under consideration. Considering the nonlinear property of integer rounding in the IDCT final stage, the newly derived formulas can estimate the evaluation results of the H.261's IDCT evaluation method. The validity of the derived formulas is verified by simulation, and it is found that they can estimate the IDCT calculation errors within the 95‐percent statistical confidence intervals and the estimation errors are within 13 percent.
Period‐adding chaos in the Si thyristor
Authors: Yasuda Yoh , Hoh Koichiro
Published: July 2007

Chaotic behaviors are observed in the anode current of the thyristors where their anodes were excited with an ac source and the constant dc current was supplied to the gate. Three aspects of chaos, i.e., period‐doubling bifurcation, intermittency, and period‐adding chaotic sequence, are observed with different combinations of control parameters.

This paper discusses primarily the feature of period‐adding chaos accompanied with intermittency. Also reported is the structure of its bifurcation by using a simple first return map model constructed from the experimental results.
Adaptive sequential testing for educational evaluation
Authors: Kobayashi Osamu
Published: July 2007

This paper proposes the adaptive sequential testing method as one means of proficiency testing. The proficiency test in general has the following problems: 1 the test contains a large number of items which are far from the educational level of the subject and deprives the subject of the motivation for the test; and 2 to secure the accuracy of the test result, the number of items is increased, which burdens the subject. To solve these problems, a system is considered in which test items for which correct/incorrect decision can be made immediately are pooled and the items are selected by a computer to be presented to the subject.

As the method of testing, the adaptive sequential testing is extended to the case of multiple hypothesis and the adaptive selection is proposed for the selection of the test items. In the proposed testing system, only the correct answer ratio for each educational level is attached to the pooled test items. By a trial simulation, the number of posed problems is halved and the system is verified to be a practical testing method with less burden to the subject.
Asynchronous simultaneous oscillations in negative resistance oscillatory circuit containing a periodically operating analog switch
Authors: Matsuki Michitaka , Mori Shinsaku
Published: July 2007

It is known that asynchronous simultaneous oscillations cannot exist in the oscillatory circuits of two degrees of freedom with a negative resistance if the current and voltage characteristics are of a third‐order polynomial and they take place as a hard oscillation if the characteristics are of a fifth‐order function [1, 2]. The authors have reported that, even in the circuit made of a third‐order negative resistance and a resonator for which no resonance occurs, a resonance occurs in a certain frequency region of switching if an analog switch is inserted between the negative resistance and the resonator and is periodically operated. It is reported that a hard oscillation phenomenon occurs in a portion of this region. It is predicted than an asynchronous simultaneous oscillation becomes possible by this hard oscillation characteristic if there are two resonators.

This paper attempts to carry out a theoretical study and an experimental confirmation for the possibility of asynchronous simultaneous oscillations by the periodic operation of a switch even if the nonlinear resistance is expressed in a third‐order function.
A computational model of auditory segregation of two frequency components—evaluation and integration of multiple cues
Authors: Kashino Kunio , Tanaka Hidehiko
Published: July 2007

As part of the basic research on perceptual sound source separation, a quantitative model is presented which gives a probability of auditory segregation of two frequency components. The model consists of independent evaluation of multiple spectral features and integration of those derived values.

First, psychoacoustic experiments which measure the probability of auditory segregation of two frequency components are described and evaluation functions of spectral features to approximate those experimental results are introduced. Then a method for integrating the spectral features is discussed and correlations between the model output and the results of the psychoacoustic experiments are measured.
Robust LQG control system under parameter uncertainty
Authors: Maki Midori , Hagino Kojiro
Published: July 2007

This paper presents a method of designing a robust linear quadratic Gaussian (LQG) controller for systems with unknown parameters when an accurate model is not available. The proposed robust LQG control scheme results in the prevention of the system performance deterioration caused by the parameter uncertainty without estimating unknown parameters explicitly.

First, an approximate model is introduced to describe the uncertain parts of the plant and a target model which the controlled system should track is constructed based on the known nominal system and the approximate signal generated by this approximate model.

Next, this tracking problem is changed to the LQG control one for an augmented system. This target trajectory is determined automatically in control operation for the augmented system. Further, both the controller and filter gain are determined as a solution of a deterministic suboptimal control problem.
Consideration of subjective evaluation method for quality of image coding
Authors: Narita Nagato
Published: July 2007

Subjective evaluations of coded images are often made by using the double‐stimulus continuous‐quality‐scale method of the ITU‐R Recommendation (old CCIR Wireless Communications Group). In this paper, the reliability of the double‐stimulus continuous‐quality‐scale method is investigated. Using this method, subjective evaluations of coded video sequences are made and the consistency of the test scores of the reference sequences is analyzed. Although it is desirable to have consistent reference test scores, results show that the test scores vary due to the influence of the quality of the video sequences being tested. As a result, dealing with image evaluation scores obtained from the double‐stimulus continuous‐quality‐scale method (with scores made on an absolute scale) may be ill advised.

Next, the reliability of data obtained from processing the differences between scores (relative scale scores) of reference and test images obtained from the double‐stimulus continuous‐quality‐scale method is investigated. Results show that compared to this method, scores obtained from the modified EBU method (modified in the sense that the video presentation method differs from the standard EBU method) have higher reliability when impairments are small and are suitable for the subjective evaluation of coded images.
Scattering center estimation of a conductive sphere using a superresolution technique
Authors: Yamada Hiroyoshi , Ogawa Yasutaka , Itoh Kiyohiko
Published: July 2007

Understanding electromagnetic scattering phenomena at high frequencies is important in specifying the location of the local scattered wave (scattering center). This procedure also is available for the practical radar target recognition. Detection of the scattering center also is possible by means of the Fourier transform in the frequency data measured by a network analyzer. However, since its resolution is limited by the measured frequency bandwidth, practical applications may often be difficult.

In this paper, the use of the MUSIC algorithm which has a higher resolution than the Fourier transform is proposed for the time‐domain estimation method of the electromagnetic scattering. As an example, the detection of the scattering center in the backscattering measurement of a conductive sphere is discussed. It is pointed out that the algorithm can be applied without significant modifications. Most of the superresolution methods require many snapshots. It is found by computer simulation that the noise suppression preprocessing by time‐domain gating is effective for reduction of the snapshots. The setting of the number of incident signals (scattering center) of the MUSIC algorithm in the time‐gated data is discussed. Further, by experiments, these investigations were verified and the high‐resolution nature of the present method is proven.
A generalized orthonormal discriminant vector method for pattern recognition
Authors: Hamamoto Yoshihiko , Tomita Shingo
Published: July 2007

Discriminant analysis on one of the well‐known feature extraction methods. Unfortunately, discriminant analysis has a serious shortcoming in that the maximum number of features is limited by the number of pattern classes. To solve the forementioned shortcoming, Okada and Tomita proposed the orthonormal discriminant vector (ODV) method based on the Fisher criterion. However, no feature is provided by the ODV method when the mean vectors are the same.

First, this paper proposes a new ODV method based on a generalized Fisher criterion which takes into consideration the difference of covariance matrices. Second, a theoretical comparison is made between the proposed method and the other feature extraction methods. Finally, experiments are conducted to show the usefulness of the proposed method.
Results of observational studies of spatial workspace collaboration
Authors: Kuzuoka Hideaki , Shoji Hiroko
Published: July 2007

Most studies on groupware have focused on desktop‐type collaboration in two‐dimensional (2‐D) space. However, to apply groupware to manufacturing or more general and free communication, groupware must be developed that supports expressions of 3‐D shapes and motions. Collaboration in the 3‐D environment will be called “spatial workspace collaboration” to distinguish it from the conventional type of collaboration.

To evaluate and understand spatial workspace collaboration, the point of focus, number of words, and verbal expressions were examined. Based on these analyses, the following requirements on groupware are deduced to support spatial workspace collaboration: realtime bi‐directional video communication; availability of nonverbal expressions; positional changeability of point of focus; ability to share points of focus; positional changeability of display; and confirmation ability.
Data maintainability of a secondary storage medium with backup copies: An evaluation model and its application
Authors: Fukumoto Satoshi , Yasui Kazumi , Nakagawa Toshio
Published: July 2007

A backup copy of a secondary storage medium is among the most practical resources in fault tolerant techniques. Data maintainability of two backup configuration types are discussed here. One is a single backup type comprising a current data medium and a backup data medium; the other is a redundant backup type comprising a current data medium and backup data two media. Taking account of state transitions of each backup configuration and the frequency of failures, a new stochastic model is proposed using a discrete‐time Markov chain to evaluate quantitatively the data maintainability of two types. The model is concretely applied to the backup configurations of floppy disks. By estimating failure probabilities of the media of floppy disks from a small‐scale questionnaire, we obtain several significant properties of two types of backup configuration for media.
Disturbances of decca waves near a huge bridge
Authors: Sato Naoto , Miyazaki Yasumitsu , Yamashita Michiko , Takenaka Yasuo
Published: July 2007

A hyperbolic navigation system using a radio wave in the 100‐kHz band is installed in the ocean surrounding Japan. This system is used widely by the fishing industry because the reproducibility of the measured location is high. Recently, near the land, several huge structures such as the Seto Bridge have been erected on the water which causes significant error in determining the location of a ship.

This paper studies the disturbances of Decca waves of the Kita‐Kyushu Decca chain due to the Minami‐Bisan Seo‐Ohashi Bridge. It was found that, with respect to the disturbance of Decca waves, the two main towers of the suspension bridge could be considered as vertical electric dipoles. In the measurement of the electrical field strength at four frequencies of the Decca station, the standing wave of the electrical field strength appeared more conspicuously with a higher frequency at the locations away from the bridge by more than 0.18 wavelength except near the bridge on the west side. It is assumed that the secondary waves from two main towers considered as vertical electric dipoles are superposed on the primary wave. Then the numerical results are carried out from the theoretical equation of the received electric field, and the measured results are explained. Also, it is found that the static capacitance of the vertical electric dipoles is 0.1 to 0.2 μF. Hence, it was possible to establish a macroscopic model of the bridge for which the main towers are considered as vertical electric dipoles.
A general method for generating odd‐shift orthogonal sequences
Authors: Ohue Kenji
Published: July 2007

In general, it is difficult to derive a finite sequence which minimizes the sidelobe of the autocorrelation function. However, it is shown that there exist the evenshift orthogonal sequences, i.e., the finite sequences for which the even‐shift autocorrelation function takes the zero value. the generation method also is shown. On the other hand, there has not been known the general method for generating the odd‐shift orthogonal sequences, for which the odd‐shift autocorrelation takes the zero value. Neither have the properties of such sequences been analyzed.

This paper presents the general method for deriving the odd‐shift orthogonal sequences, using the complex function as the generating function and constructing the numerical sequence from the Fourier expansion coefficients. First, the one‐ and two‐dimensional odd‐shift orthogonal complex sequences and odd‐shift orthogonal real sequences are defined from their autocorrelation functions. Then, it is shown that those numerical sequences are odd‐shift orthogonal sequences if they are skew‐symmetrical sequences. Finally, the properties of the generating functions that derive those odd‐shift orthogonal sequences are shown, together with the generation algorithm.
FIR and IIR SC interpolation filters using parallel cyclic‐type circuit for high frequency
Authors: Hashimoto Mitsuru , Hirata Yoshinori , Takahashi Nobuaki , Takebe Tsuyoshi
Published: July 2007

High‐frequency responses of switched‐capacitor filters deteriorated mostly as a result of the finite gain bandwidth product of an operational amplifier. the authors previously proposed a type of FIR and IIR switched‐capacitor filter using a parallel cyclic‐type circuit in order to improve high‐frequency performance. In the parallel cyclic‐type circuit, an operational amplifier is not used as a delay element. Instead, both delay and weighting operations are implemented with switches and a coefficient capacitor simultaneously. Only one operational amplifier is used for a summing operation. Consequently, the maximum operational frequency of the filter is extended up to the unity gain frequency of the operational amplifier.

This paper proposes a new type of FIR and IIR switched‐capacitor interpolation filter that can be operated in a high‐frequency region by using the parallel cyclic‐type circuit. Effects of both the finite gain bandwidth of the operational amplifier and parasitic capacitances on filter performance are analyzed by applying the analysis method of periodically time‐varying digital filter. These effects are found to be very few in number below the unity gain frequency of the operational amplifier. Third‐order FIR and biquadratic IIR interpolation filters are constructed with discrete components and verified to have good frequency performance that coincides with the analyzed one.
Systematic construction algorithm of fixed block‐length multitrack (d, k) Codes
Authors: Tanaka Hatsukazu
Published: July 2007

This paper notes the fact that the recording code often takes the form of a multitrack code, as well as the fact that the k‐constraint in the (d, k)‐constraint is considerably relaxed, which increases the capacity. Then the paper proposes a systematic construction algorithm for the fixed block‐length multitrack (d, k) code and discusses the coding characteristics. A simple idea is applied in the code construction to apply the (d, ∞) code to all tracks except for one in order to reflect the relaxation of the k‐constraint on the improvement of the coding rate. the (d, ∞) code is applied to the left track if the vector obtained by the elementwise logical sum of the code words satisfies the k‐constraint and, if not, the (d, k) code is applied. the author has already proposed a systematic construction algorithm for the efficient fixed block‐length (d, k) code.

This paper discusses mainly the systematic construction algorithm and the coding characteristics for the fixed block‐length (d, ∞) code, based on the unique ordering technique of the binary sequence by the Schalkwijk algorithm, as the systematic construction algorithm for the fixed block‐length multitrack (d, k) code based on the forementioned idea. the coding rate and the coding efficiency are examined, and the result indicates that the characteristics are very close to those of the (d, ∞) code.
Asymptotic capabilities of concatenated codes and iterated codes
Authors: Nishijima Toshihisa , Hirasawa Shigeichi
Published: July 2007

As the codes that can prove Shannon's channel coding theorem by constructing the code, only two classes of codes are known, i.e., the concatenated code proposed by Forney and the iterated code proposed by Elias. In spite of this fact, those two codes have not been compared by the common evaluation measure. the former is evaluated from the standpoint of information theory and coding theory in terms of the average power from the viewpoint of the reliability function and the asymptotic distance ratio of the constructed code. For the latter, it is shown under a restricted condition that the decoding error rate per binary symbol converges to zero, only for the constructed code.

With this as background, this paper compares the concatenated code and the iterated code by the common evaluation measure, considering the computational complexity required for the decoding. It is shown that the concatenated code is better than the iterated code. It is shown also that those two kinds of codes can be realized with the decoding complexity of polynomial order of the code length. Under the foregoing condition, it is shown that it is impossible to construct an iterated code with the nonzero asymptotic distanc ratio.

For the Justesen code, which is a class of the constructive concatenated codes, and the Elias iterated code, which is a class of the iterated codes, the upper bound for the decoding error probability is separated into a function of code length and a function of code rate. the code length is represented as a function of the computational complexity required for the decoding, and the upper bounds of the decoding error probabilities of the two codes are compared for the same computational complexity and the code rate. It is shown as a result that the upper bound for the decoding error probability of the Justesen code is asymptotically better than that of the Elias iterated code.
A method for predicting chaotic time‐series with outliers
Authors: Itoh Ken‐Ichi
Published: July 2007

In recent years, there have been intensive studies on the short‐term prediction method for time‐series data with chaotic properties. the prediction for the chaotic time‐series data is executed, for example, as follows. Based on the single‐variable time‐series data, the trajectory of the attractor is reconstructed in the multidimen sional space and the change of the trajectory is predicted by a polynomial approximation. However, when there exists a noise such as the outlier in the time‐series, the prediction error is increased drastically.

To solve this problem, this paper proposes a prediction method which is robust against the outlier. More precisely, the following three elaborations are added to the conventional method: (1) the coefficients of the polynomial are calculated by Biweight's estimation method; (2) the order of the polynomial is determined automatically examining the residual error; and (3) the data with a large prediction error are replaced by the predicted value and the next prediction is applied. an evaluation experiment is executed for the data where the pseudorandom variable with the Cauchy distribution is superposed on the chaotic time‐series data generated by the Hènon map and the Lorenz model. It is verified as a result that the proposed method can well suppress the increase of the prediction error due to the outliers, improving greatly the accuracy of the prediction compared to the conventional method.
Design of a wide bandpass filter with attenuation poles using lines and capacitances
Authors: Sekine Toshikazu , Nakamura Takashi , Yokokawa Senji
Published: July 2007

This paper presents a design of a compact wide bandpass filter composed of transmission lines and capacitances having attenuation poles in high frequen‐cies. The design method uses a frequency transformation and some approximations to the ladder‐type network obtained from a circuit transformation of the transmission line into a π‐ type network. As a result, this filter design is simple. The influence of the approximation to the attenuation characteristics of the filter is shown. The experimental results show that the design of this filter is effective.
“Hues” sound level meter based on noise‐rating scale composed of five‐hue scale and equivalent continuous a‐weighted sound pressure level and its usefulness
Authors: Furihata Kenji , Yanagisawa Takesaburo
Published: July 2007

The noise level and the effective noise level used for noise evaluation are physical indices and generally are not understood. It is believed that the understanding of such physical indices can be enhanced if one can realize a sound level meter which can display directly the relationship of the psychological effect by the noise with the measured decibel scale.

From this point of view, this paper first evaluates in a laboratory experiment the relationship between the hue level expressing the psychological effect and the effective sound level. According to the obtained noiserating scale, the five‐hue scale, “red, purple, yellow, green, blue” and the effective noise level correlate well.

Next, a new sound level meter including the obtained noise‐rating scale (to be called the hue sound level meter) and the existing sound level meter displaying only the measured value were compared at the actual measurement site. From the comparison, it was found that the hue sound level meter has better visual display and the display represents the auditory sense. From the foregoing results, it is possible to promote general understanding of the physical index.
An adaptive α‐trimmed mean filter with excellent detail‐preservation and evaluation of its performance
Authors: Taguchi Akira
Published: July 2007

The nonlinear filter is useful in restoring an image which is degraded by additive noise. the α‐trimmed mean filter is a generalization of the median filter. In the α‐trimmed mean filter, the data within the filter window are examined and a specified number of samples is selected as close to the median value from the ordered statistical viewpoint. Then the mean of the selected samples is output. This mechanism is powerful in eliminating the impulsive noise and suppressing the non‐impulsive noise. On the other hand, it should be noted that the image and other signals are composed of various components (such as the flat part and the edge) and it is indispensable that the number of samples to be averaged in the α‐trimmed mean filter should be made variable according to the local information (i.e., adaptive filtering) to realize a highly precise restoration of the signal.

From another aspect, the α‐trimmed mean filter has a problem in that the detailed behavior of the signal cannot be preserved when the filter window is large. From such a viewpoint, this paper newly proposes an adaptive α‐trimmed mean filter based on the local statistics, which has an excellent performance in the preservation of the detailed signal components. the performances in the preservation of the edges and the elimination of the noises are evaluated.

Finally, the filter is actually applied to the image restoration and the practical usefulness of the proposed filter is demonstrated.
DC component of the stops for speech training
Authors: Aoyama Kensei
Published: July 2007

The expired air used in the generation of speech is released with the speech and produces a unipolar pressure variation of the air. This pressure fluctuation is considered as a kind of speech and is called the dc component. the dc component is composed of the expired air affected by the articulation organ and is considered to contain the articulation information.

This study aims at the use of the dc component in the utterance training. Using the variable reactance pressure sensor, the relation between the dc component of the stop and the articulation is examined. the following results are revealed.

The dc component of the stop is radiated from the mouth with a sharp directivity specific to each stop. the dc component of the stop is a sharp triangular pulse wave, with the peak value (i.e., the maximum of dc component) and the peak position related to the articulation position and the utterance intensity. In order to learn the utterance of the stop, it is required to master the correct articulation, especially the articulation position, but it is fairly difficult to determine the articulation position from the speech wave. On the other hand, it is easy to estimate the articulation position from the peak value and the peak position of the triangular pulse of the dc component, together with the direction of radiation of the dc component. In other words, it is seen that the dc component can be utilized in the utterance training.
Measuring the density of snow particles and snowfall rate
Authors: Muramoto Ken‐Ichiro , Matsuura Kohki , Shiina Toru
Published: July 2007

It is important to measure the quality of falling snow particles as a help in analyzing radio attenuation during snowfalls. Since the quality of snow particles depends almost entirely on their density (the amount of water they contain), the snow particle density reflects their quality. to permit automatic measurement of the average densities of falling snow particles over long periods of time, the diameters and fall velocities of snow particles are measured by means of image processing. Simultaneously, the weight of all snow particles that have fallen to the ground was measured directly by means of an electronic balance. the density of the falling snow particles is calculated by dividing into this total weight the total volume of snow particles that passed through a unit volume (obtained from the diameter and velocity of snow particles). the diameters, velocities, and number of snow particles are determined by image processing. Snowfall rates calculated from the image processing data are compared with those measured directly and are found to agree.
Complex coefficient reference networks with transmission zeros on imaginary axis
Authors: Yamazaki Osamu , Takahashi Shin‐Ichi
Published: July 2007

This paper considers the complex coefficient transfer function with transmission zeros on the imaginary axis and presents a method to realize such a transfer function by a reactance network including imaginary resistors with double resistor termination. Using the obtained network and the reference network for the complex‐wave digital filter, the complex digital filter with transmission zeros on the imaginary axis can easily be extended, which is important in practice.

In this paper, Brune's method modified by Belevitch as well as Fujisawa's method extended by Morimoto et al. are improved and the realizability condition is relaxed. Then, several construction examples are shown and the features of construction methods are compared and discussed.
Filter cirucit realization with single operational amplifier and LCR elements and its application for allpass response
Authors: Hirano Satoshi , Hayahara Etsuro
Published: July 2007

This paper discusses a method for configuration of a single operational amplifier filter circuit using one operational amplifier (Op Amp) and LCR elements. the proposed single Op Amp filter cirucit has a constant resistance input impedance and a zero output impedance. It allows realization of any stable transfer function.

In this paper, a method for realizatio of an allpass function in investigated in particular and its design method is presented. Actually, a computer simulation and an experiment with a test circuit made of discrete components are carried out for a cirucit to realize a fifth‐order allpass function. the cirucit performance is compared with that of a cascaded scheme of Deliyannis circuits. the element sensitivity in regard to the LCR elements, the frequncy characteristics of the Op Amp and the sensitivity of the inductor L with respect to Q are derived. It is shown that the proposed allpass circuit has low element sensitivity and the number of necessary reactance elements is identical to the order of the transfer function.
Improved cedervall‐Johannesson algorithm for computing the distance spectrum of convolutional codes
Authors: Morii Masakatu , Tanigawa Kouichi , Sasano Hiroshi
Published: July 2007

In the convolutional code, the minimum free distance is the important parameter to evaluate the performance of the code. Recently, it has been considered necessary, however, for a more strict evaluation of the performance to examine the number of code words with weights around the minimum free distance, i.e., the more general distance spectrum. In general, it is difficult to derive the distance spectrum of a code, including the derivation of the minimum free distance when the code size is enalrged, and it will be significant if it is derived easily.

This paper proposes an algorithm that can derive the distance spectrum with a high speed, compared to the method already proposed. The algorithm presented in this paper is an improvement of the fast algorithm for searching trees (FAST) proposed by M. Cedervall and R. Johannesson. The distance spectrum can be derived with approximately one‐half the computational complexity compared to FAST.
A median filter based on fuzzy rules
Authors: Arakawa Kaoru
Published: July 2007

This paper proposes the median filter based on the fuzzy rules, as a filter that effectively eliminates the impulsive noise without greatly distorting the signal such as image. It is already known that the median filter is effective in eliminating the impulsive noise, but the median filter also distorts the signal waveform itself.

To cope with this situation, it is desirable that the filter should operate as the median filter when there exists the impulsive noise and, if not, should only pass the input to the output. In this approach, however, the rule to decide on the existence/nonexistence of the impulsive noise generally is ambiguous.

From such a viewpoint, this paper proposes a new type of median filter where the above decision rule is represented by the fuzzy rules and the output is controlled by the result of the decision. the filter is designed optimally by training. the effectiveness of the filter is demonstrated by actually eliminating the noise from the image.