bluuebunny/crossref_metadata_2025_split huggingface dataset:
Systematic construction algorithm of fixed block‐length multitrack (d, k) Codes
Authors: Tanaka Hatsukazu 
Published: July 2007

AbstractThis 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

AbstractAs 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

AbstractIn 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

AbstractThis 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

AbstractThe 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

AbstractThe 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

AbstractThe 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

AbstractIt 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

AbstractThis 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

AbstractThis 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

AbstractIn 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

AbstractThis 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.
An infinite impulse response adaptive digital filter for line‐canceller and its pole‐control algorithm
Authors: Nakanishi Takahiro , Takumi Ichi , Hata Masayasu , Yamada Kouichi 
Published: July 2007

AbstractThis paper proposes the parallel construction of the second‐order infinite impulse response (IIR) adaptive digital filter (ADF) as an adaptive digital filter for eliminating multiple narrowband signals mixed in the broadband signal. Then the pole‐control method for the filter is proposed. In the propsed pole‐control, the pole radius and the pole argument are controlled separately by different algorithms.In this study, the pole argument is corrected by estimating the deviation of the central frequency of the signal to be eliminated from the convergent value of the multiplier coefficient that determines the zero. the pole radius is controlled by the method that determines the direction of correrction, using the parallel‐connected three ADFs with the same pole argument and a slightly different pole radius. It is verified by a simulation that those pole‐control methods work effectively. It is demonstrated that the pole radius and the pole argument can be controlled simultraneously.
Problems of where to locate p‐sinks in a flow network
Authors: Watanabe Kaoru , Sengoku Masakazu , Tamura Hiroshi 
Published: July 2007

AbstractAs location problems in flow networks, there are p‐center and p‐median problems; and they are known to be solvable in polynomial time. In this paper, we propose a new location problem. It is the problem of locating p‐sinks so that the maximum flow is maximized given a flow network N with a source (this problem is called a p‐collection problem). First, a linear time algorithm to solve the 1‐collection problem for trees is given. Then a pseudo‐polynomial time algorithm to solve the p‐collection problem for trees based on dynamic programming is presented. Since the decision problem corresponding to the p‐collection problem for trees is known to be NP‐complete, its decision problem turns out to be strongly NP‐complete.
Optimal garbage collection policies for a database with random threshold level
Authors: Satow Takashi , Yasui Kazumi , Nakagawa Toshio 
Published: July 2007

AbstractWhen a database is updated, it is difficult to avoid divided accumulations of data in storage areas by adding and deleting them. Such divided accumulations decrease access efficiency; hence, efficient maintenance is needed to meet the requirement of high‐quality service to users. However, maintenance costs must also be kept low. However, if the operating time of a database is specified, economical and planned maintenance to guarantee that high‐quality services can be implemented.This paper considers the following stochastic model: A database is updated in accordance with a stochastic process and garbage is generated in the time between updatings, causing fragmentation in storage areas. Maintenance costs increase when accumulated garbage exceeds a threshold level. That is, to remove fragmentation from storage areas, garbage collection is made at time T or at the N‐th update. the expected cost rate of this model is obtained and optimal time T* and number N*, which minimize it, are derived. Furthermore, numerical examples are given and some discussions for these examples are presented.
Obtaining a marked graph from a synchronic distance matrix
Authors: Tamura Hiroshi , Abe Takeo , Saitoh Satoshi , Mikami Kiyoshi , Sengoku Masakazu , Yamaguchi Yoshio , Shinoda Shoji 
Published: July 2007

AbstractThe notion of synchronic distances is a fundamental concept introduced by Petri [1]. It is a metric closely related to a degree of mutual dependence between two events in a condition/event system. the synchronic distance between two vertices in a marked graph is defined here as the maximum number of times one vertex can fire without firing the other. Marked graphs form a subclass of Petri nets. the synchronic distance matrix of a marked graph M with n vertices is an n × n symmetric matrix D* = [dM*(vi, vj)], where dM*(vi, vj) is the synchronic distance between vertices vi and vj. Conversely, using matrix D, the focus is on finding a marked graph whose synchronic distance matrix is D. Murata et al. [2] show that the synchronic distance matrix of a marked graph is a distance matrix of an undirected network and give a method for finding a marked graph when D is realizable as the distance matrix of a tree with positive integer edge weights. This paper presents a necessary and sufficient condition for D to be the synchronic distance matrix of a live marked graph. A matrix D is given such that D is a distance matrix of an undirected network and there does not exist a marked graph whose synchronic distance matrix is D. Also, a method is proposed for finding a live marked graph when D is realizable as the distance matrix of a tree as well as that of a nontree graph like a cycle.
Decoding of reed‐solomon codes beyond the bch bound using euclidean algorithm
Authors: Horiguchi Toshio 
Published: July 2007

AbstractThe output polynomials of Euclid's decoding method for t‐error‐correcting Reed‐Solomon (RS) codes with design distances of d = 2t + 1 and d = 2t + 2 are analyzed when t + v (v ≥ 1) errors occur. A method is proposed which expresses the error‐locator polynomials for t + v errors in terms of output and undetermined polynomials. the error‐locator polynomials are expressed for v = 1, 2 as specific examples that are important in applications. Furthermore, a decoding algorithm is proposed that corrects t + v errros, which exceeds the BCH bound, by estimating the undetermined polynomials. the proposed algorithm requires O(2v ‐ 1n2vlog n) arithmetic operations for decoding t + v errors with a t‐error‐correcting RS codes with length n and a design distance of 2t + 1, whereas previous decoding methods that exceeded the BCH bound required O(n2v + 1log2 n) operations. Also, it is shown that the proposed decoding algorithm can be applied to continued fraction decoders.
A method for quantizing coefficients of a filter with genetic algorithm
Authors: Aketa Yoshihiro , Haseyama Miki , Kitajima Hideo , Nagai Nobuo 
Published: July 2007

AbstractThis paper proposes a new method to quantize coeficients of filters where frequency‐characteristic deteriortion is suppressed. When a digital filter is realized, quantization errors cannot be avoided because operation circuits and registers have finite word lengths. Generally, filter coefficients are quantized by popular methods, such as rounding and omitting, etc. The proposed method obtains the quantized coefficients by search. The Genetic Algorithm is used for the search because it needs only objective function information. Further, by using the proposed method, the filter coefficients can be quantized with the frequency weight. Therefore, deterioration of the coefficients quantized by the proposed method is suppressed in the specified frequency region. The effectiveness of this method is shown by some simulations that quantize the four‐line lattice and the direct form filter coefficients.
Burst error correcting binary cyclic an codes consisting of 2m ‐ 1 codewords
Authors: Murakami Ryutaro , Okura Yoshiteru , Shimada Ryosaku 
Published: July 2007

AbstractArithmetic an codes are useful for detecting and correcting errors in arithmetic operations and digital data transmission. Especially, cyclic an codes are considered to be useful. First, outlines of binary cyclic an codes and burst errors are described. Second, a method to construct the binary cyclic an codes consisting of 2m ‐ 1 codewords is presented and the burst error correcting ability of these codes is discussed. the number of codewords generally is one of the most important parameters for selecting a suitable code for a given system. From these codes, a binary cyclic an code satisfying requirements of both the number of codewords and the correcting ability in the system can easily be chosen. A relation between the code rate and the correcting ability also is discussed. Finally, another relation between the burst error correcting ability and the random error correcting ability is discussed. As a result, using the constructing method proposed here, binary cyclic an codes can be constructed which will be useful not only for the burst error correction but also for the random error correction.