bluuebunny/crossref_metadata_2025_split huggingface dataset:
Conditions of convergence with respect to extended forms of interpolatory approximation for multidimensional waves
Authors: Kida Takuro 
Published: July 2007

AbstractIn this paper we consider a family of output waves obtained by impressing simultaneously an n‐dimensional input wave f(X) to the finite number of time‐invariant linear networks. A problem is considered, where the original wave f(X) is to be approximated using the sampled values of the forementioned output waves. A unified theory is presented with respect to the convergence of the approximate wave to the original wave f(X). the discussion given in this paper may be considered as a basis for the future theory of subband coding and multiplexing of the multidimensional signals, in the sense that we use the sampled values of the multidimensional waves transferred through the linear networks. the approximate wave h(X) for f(x) is a sum of the forementioned sample values of these output waves, being multiplied by certain n‐dimensional waves called interpolation functions. the set of sampling points considered in this paper includes most of the practically important arrangements of sampling points, such as hexagonal and octagonal lattices on the two‐dimensional plane. the set of f(X) in this paper is defined as the set of n‐dimensional waves such that the corresponding Fourier spectrum has the weighted power‐of‐p norm (p > 1) which is less than a given positive value. Under several assumptions, sufficient, necessary, and necessary and sufficient conditions are derived for the wave h(X) always to converge to f(X). These convergence conditions are presented which are imposed on the transfer functions of the time‐invariant linear networks through which f(X) propagates as well as the arrangement of the sampling points and the interpolation functions. They are closely related to the rank of the system of linear equations determined by those parameters.
Recognition of vowels and a semivowel using formant locus extracted by cubic spline
Authors: Matsumura Masafumi , Yamane Hiroaki , Fujii Katsuhiko 
Published: July 2007

AbstractThe purpose of this study is to recognize the semivowel /j/ and vowels from the continuous speech of an unspecified speaker. the format frequencies are used as the feature parameters, which correspond to the articulation by the tongue and the lower jaw. In the extraction of the formant locus, the continuity is emphasized. A pre‐processing is made which detects the missing formant or unnecessary peaks of the speech spectrum. Then the interpolation and smoothing are made using the spline function. A triangle on the formant plane F1‐F2 plane) is considered, where the vertices represent the articulations of /a/, /i/ and /u/. the triangle is transformed into an equiliateral triangle. By this transformation, the speaker differences of the positions of the vowels on the F1‐F2 plane are normalized. the point minimizing the rate of change of the normalized formant frequency is determined, and, based on the result, the segmentation and recognition are performed. the recognition rate of 92.8 percent was achieved for the vowels in the continuous speech uttered by ten speakers. Then based on the analysis of articulation by tongue and lower jaw in pronouncing the semivowel /j/, a recognition experiment was attempted by extracting the feature parameters for discrimination from the shape of the formant locus. As a result, the semi‐vowel recognition rate of 94 percent was achieved for continuous speech uttered by nine speakers.
A 1024 × 1024 channel‐switching lsi and its application to switching systems
Authors: Amada Eiichi , Kuwahara Hiroshi , Shirasu Hirotoshi , Shibukawa Masaru , Ikuzaki Kunihiko , Harakawa Takao 
Published: July 2007

AbstractThis paper presents a configuration of switching LSI based on 1.3 um CMOS technology, its function and application example to the channel switching. the switching LSI has a function to switch arbitrary time slots of the input and output highways. the switching capacity of 1024 × 1024 channels was realized by a single chip. the LSI includes the higher‐level functions of processor interface, but frame synchronization and fault monitoring. As an example of applications of this LSI, a switching network with 32768 channels (max) is realized.
Construction of variable cutoff digital filter based on a changeover of taps
Authors: Ikehara Masaaki , Takahashi Shin‐Ichi , Kato Mariko 
Published: July 2007

AbstractThis paper shows that by taking every M‐th impulse response from the impulse responses of FIR low‐pass digital filters and constructing a new digital filter based on them, a filter is obtained with low‐pass characteristic such that the passband width is M times that of the original filter. Next, taking advantage of this characteristic, a method is proposed for designing a low‐pass digital filter with variable cutoff frequency based on the same circuit by first designing a basic digital filter and varying M, that is, changing over taps. This principle is applicable not only to FIR filters but similarly also to IIR filters if some restrictions are allowed for denominator polynomials. Further, it is shown that by applying this theory to IFIR (Interpolated FIR) digital filters and multirate digital filters, filters can be realized which are more effective with respect to their cutoff characteristic and operational points.
A layout design methodology using data description with correspondence between circuit and layout
Authors: Okuda Ryosuke , Onodera Hidetoshi , Tamaru Keikichi 
Published: July 2007

AbstractProblems at present in the manual layout design of logic LSI are the high desing cost due to the overhead of detection and correction of design errors, as well as the difficulty in modifying and reusing the design data. to solve these problems, this paper proposes a data representation providing the correspondence between the circuit structure and the layout, together with a new layout design method using that idea. This data representation leads to a unified treatment based on the correspondence between the circuit data and the layout data. It is made easy to modify the design and to maintain or reuse the design data. the layout design method based on the proposed idea effectively utilizes the circuit data in the layout design, which can completely prevent routing errors, and simplifies various layout verifications. Further, this paper presents an example wherein the proposed layout design method is applied to the symbolic layout system for MOS logic LSI. the result of layout experiments indicates the usefulness of the proposed design method.
High‐speed decoding of BCH codes using a new error‐evaluation algorithm
Authors: Horiguchi Toshio 
Published: July 2007

AbstractA new error‐evaluation method for computing error values in decoding Reed‐Solomon codes or nonbinary BCH codes using the Berle‐kamp‐Massey algorithm is proposed.Since the error‐evaluator polynomial used in the new method is obtained as by‐products of the computation for the error‐locator polynomial, the new method is more efficient than Forney's method whose error‐evaluator polynomial cannot be obtained as the by‐products of the error‐locator polynomial.
Analysis of sound fields in rooms using Bergeron's method
Authors: Shimoda Hidemaro , Yoshida Norinobu , Fukai Ichiro 
Published: July 2007

AbstractAt the designing stage of an auditorium, e.g., a concert hall or theater, it is imporant to predict its acoustic field via a computer. A conventional prediction method based on geometrical acoustic theory gives only a macroview of the field, and an addirional scale model test is required for a further investigation. to overcome this problem, although a technology to treat the sound field in a room as a wave field has been considered, there have been few examples of hree‐dimensional time‐response solution of the sound field in a room because of the difficulty of handling a wide analytical region, ide frequency band, and complex boundary conditions. This paper examines the justification of application of Bergeron's method to three‐dimensional sound field in a room. This paper also analyzes the correspondence between a three‐dimensional equivalent‐ciruit representation and the sound field, and examines the fundamental nature of a free sound field. Based on the results, a model of a simple rectangular room was considered, and its steady‐state sound pressure distribution (which is a fundamental characteristic the field) and simulation of reverberation (transient phenomenon) were examined. All the successful results show the usefulness of the proposed method.
An analysis method for acoustic waves composed of cosine waves
Authors: Tamaribuchi Kazuhiro , Saito Shuzo 
Published: July 2007

AbstractThis paper proposes a method of wave analysis which determines the frequency, initial phase and amplitude of the frequency components from three times the number of data compared with the number of components, using only the algebraic method. In the analysis, the frequencies are determined using the formula front sum to product of trigonometric functions, assuming the waveform is composed of a finite number of frequency components. Then using the determined frequencies, the initial phases and the amplitudes are determined. For verification, the signal of the in‐band multifrequency signaling system is considered as an example, and a computer simulation was made for the proposed frequency analysis. the signal is composed of one each from four kinds of low and high frequencies. Sixteen kinds of composite waves are constructed. the analysis is made as follows. Variations of ±1.3 percent are given to each of the low and high frequencies, and the initial phase was set as arbitrary in the range from ‐90 to +270 deg. in regard to the cosine wave. the amplitude is varied from 1 to 1/16. the sampling frequency is 4.48 kHz and the amplitude quantization is made in 12 bit. the maximum error in the result of analysis for the multiple frequency signal is ‐3.9 percent for 100, 000 trials, including the given frequency shift. In the analysis of wave composed of five components, the accuracy of nine digits is realized, using the double precision discrete data.
Equiripple magnitude approximation for low‐pass IIR digital filters with prescribed group delay
Authors: Nishikawa Kiyoshi , Choi Changseok 
Published: July 2007

AbstractAn efficient algorithm is proposed for approximating the magnitude of the low‐pass IIR digital filter with prescribed group delay in equiripple. Prescribed group delay characteristics can be realized by placing in the denominator of the transfer function a polynomial expression designed to have such characteristics. For amplitude characteristics we approximate it to obtain equiripple using a linear‐phase polynomial in the numerator.Equiripple magnitude approximation is done separately for the passband and for the stopband and all the characteristics are made equiripple by several times of iterations.In this simple method the auxiliary complex plane is utilized and in each the transformed transfer function is expressed as the sum of objective magnitude 1 and the error function for it. By assigning an all‐pass function to an approximation function of the error function, equiripple characteristics are realized.By approximation in the stopband, good equiripple characteristics can be obtained even if all the degrees of freedom are used. On the other hand, since the approximation in the passband deviated largely from equirippleness, a method is presented for improving ripple characteristics by reducing the degree of freedom by one. As a design example, a filter of order 10 is given.
A multilevel data transmission baseband LSI using SC circuits
Authors: Nakayama Kenji , Ichihara Masaki 
Published: July 2007

AbstractThe baseband circuit for multilevel data transmission system was implemented by LSI using SC circuit technology. The waveform‐shaping filter, the full‐wave rectifier for clock regeneration, the narrowband filter and other circuits were constructed by SC circuits. The size and the power dissipation were reduced by constructing all filters in time‐division multiplex. The chip area and the power dissipation in the SC circuit is proportional to the total capacitance, and the total capacitance is determined by the capacitance ratio and the unit capacitance.From such a viewpoint, the narrowband filter used in the clock regeneration is constructed by multirate, and the capacitance ratio was compressed using the N‐path filter. Capacitance partitioning is applied to high‐Q filters to suppress the capacitance ratio. Then by composing all capacitances as the integer multiples of the same unit capacitance, the desired ratio accuracy was achieved based on the unit capacitance of 0.1 pF. The problem of performance degradation due to integer capacitance ratio was minimized by the discrete approximation.When the N‐path filter is composed by the time‐division multiplex scheme, the performance may be degraded greatly due to the parasitic capacitance among wirings. This effect was analyzed and an improvement from the viewpoint of the layout was proposed. As a result, the multilevel data transmission baseband LSI was realized with the chip size of 4.3 × 4/2 mm2 and the power consumption of 35 mW.
Numerical simulation of characteristics of condenser microphones at high ambient pressure
Authors: Suzuki Hisayoshi , Uchida Yoshiyuki , Oohashi Masanao , Nakai Takayoshi 
Published: July 2007

AbstractThe characteristics of a microphone are affected by the pressure and composition of the ambient gas. This paper presents a numerical simulation of the microphone within nitrogen gas with various pressures up to 48 atm. The simulation was carried out both for the deflection (static deflection) of the membrane by the polarization voltage and for the vibration (dynamic deflection) of the membrane by the incident acoustic pressure. In these cases, the physical constants of the gas, the location of the hole on the back plate and the gas flow between the back plate with a hole and the membrane were considered. The case also was considered in which the location of the hole is axially nonsymmetric. It was found that the membrane vibration derived by simulation corresponds well with the experimental result on the actual microphone as reported previously. From these simulations, the sensitivity frequency response of the condenser microphone under a high‐gas‐pressure environment were better understood.
Chaotic phenomena in an auto‐gain‐controlled oscillator using a pulsewidth controller
Authors: Uehara Tetsuo , Mori Shinsaku , Inaba Naohiko 
Published: July 2007

AbstractThis paper shows that in a certain type of amplitude‐controlled oscillator using a pulsewidth controller, correspondence of the Poincaré map with the cut map and logistic map can be obtained. It is proven almost analytically that the Poincaré map is ergodic if it becomes a cut map. By a computer‐assisted analysis, it is proven that chaos in the sense of Li‐Yorke is generated if it is a logistic map. Further, the validity of the analysis is confirmed by a circuit experiment.
Synthesis of LC simulation‐type RC active filters with reduced number of voltage followers
Authors: Ishibashi Yukio 
Published: July 2007

AbstractIt is known that the active RC filter which simulates the LC ladder filter has lower element‐sensitivity. However, recently, many such types of filters have been proposed. In this paper, a method of realization by low‐pass filter simulation‐type active RC circuit is described.First, it is proposed that an LC filter to be simulated be realized by adding resistors in parallel to L's and in series to C' s of the conventional LC filter. Such a filter has the transmission zeros on the negative real axis of the 5‐plane, thus the conventional transfer function can no longer be used.Next the derivation method of the transfer function for this filter will be presented. Then the realization of the LCR filter obtained by the active RC circuit using a reduced number of voltage followers will be described.Finally, the synthesis and experiment of a fifth‐order Chebyshev low‐pass filter will be given. The effectiveness of the synthesis method is confirmed by the fact that the analysis of the element sensitivity by computer shows that in the passband it is below about 50 percent that of the leapfrog type circuit.
Maximum likelihood decoding of convolutional codes using channel states
Authors: Saito Osamu , Imai Hideki 
Published: July 2007

AbstractViterbi decoding and sequential decoding are well known as the maximum or approximate maximum likelihood decoding methods for the convolutional code. The decoding error rates of those decoding methods have been examined in the past. However, in those discussions the evaluation has been made by assuming the channel as memoryless. By contrast, this paper proposes several maximum likelihood decoding methods for the convolutional code in the channel model with memory, such as the Gilbert model. In decoding method (I), the conditional probability of the transmitted sequence for the received sequence is determined for each state of the channel. Using the result as the metric, the state of the channel is considered in the same way as the code trellis. The decoding is made by suitably selecting the metric. In decoding method (II), the state of the channel is estimated to suppress the computational complexity with the increase of the channel states. These methods are applied to Viterbi decoding. It is shown by computer simulation that the decoding error rate is improved over the traditional Viterbi decoding with equal complexity in instrumentation compared with the traditional decoding methods.
Fundamental studies on combined coding for information source and channel
Authors: Fukunaga Tetsuya , Murata Masashi , Kasahara Masao 
Published: July 2007

AbstractFrom both theoretical and practical viewpoints, one of the most basic and important problems in the digital communication system is to minimize the distortion between the output signal from the information source and the destination signal on the receiving side, under the condition that the transmission rate in the channel is fixed. From such a viewpoint, this paper presents a detailed investigation of the trade‐off relation between the vector quantization (source coding) and the error‐correcting coding (channel coding), where the transmission rate of the channel is fixed as constant. The combined source and channel coding is shown, which is optimal in the sense that the average overall distortion is minimized in the communication system. The minimized average distortion is compared with the average overall distortion, where the ideal error‐correcting code with the minimum distance given by Hamming's upper bound is applied. A detailed discussion is made. The memoryless Gaussian source, the squared‐error distortion measure, the binary symmetric channel and the binary (n, k) block error‐correcting code are assumed. Finally, an investigation is made using the actual speech communication system.
CAM‐based hardware engine for geometrical problems in VLSI design
Authors: Suzuki Kei , Ohtsuki Tatsuo 
Published: July 2007

AbstractA new hardware engine called CHARGE (CAM‐based Hardware Engine for Geometrical Problems) is presented to accelerate basic geometrical search algorithms for VLSI layout design and verification. Due to the world‐parallel nature of the Content Addressable Memory (CAM), basic geometrical search problems can be solved in constant time, independent of the number of geometrical data. This approach has flexibility to exclude complicated coding for various data structures depending on subproblems. The architecture of CHARGE and performance data measured on a prototype machine are described.
Design of three‐dimensional digital filters based on the outer product expansion of impulse response specifications
Authors: Ohki Makoto , Kawamata Masayuki , Higuchi Tatsuo 
Published: July 2007

AbstractIn dynamic image processing, medical image processing and seismic signal processing, high‐level processing, which cannot be executed by 1‐D or 2‐D signal processing, is expected to be realized by 3‐D digial signal processing. This paper proposed an efficient design method of 3‐D digital filters by the 3‐D outer product expansion which was proposed to decompose 3‐D arrays in image coding. In the proposed method, the given specification for the 3‐D filter is decomposed into the specifications for 1‐D filters using 3‐D outer product expansion and reducing the design of the 3‐D filter to that of the 1‐D filters. When the given impulse response specification is symmetrical, the computation required for the design in the proposed method can be reduced.
Analysis of parallel redundant system with dependent failures
Authors: Hidaka Toshimitsu 
Published: July 2007

AbstractIn communication line systems, in general, failure correlation (dependency of communicational line (path) composing a system due to the existence of common‐caused failure mode) often occurs and dependent failure correlation between communication lines (paths) must inevitably be considered in many cases. A concrete example is in reliability analysis between communication terminals in communication cable lines linking two central terminals, a characteristic quantity analysis of reliability when r‐fold common‐caused failures of neighboring lines occur. Thus, in this paper, we take a parallel redundant system given by common‐caused failure (r) and the number of communication lines (n), etc.
A bayesian reliability analysis model by a single debugger for discrete‐type software and its application
Authors: Sandoh Hiroaki , Fujii Susumu , Yamawaki Tatsuya 
Published: July 2007

AbstractA reliability analysis model is proposed for software products at their development stage from the Bayesian viewpoint, with emphasis on such software products as the usual mathematical software used discretely in time. In the proposed model, a single debugger is engaged in conducting software tests and in debugging a software product, where the significations of “single debugger” are as follows: the model does not consider the case where several software engineers conduct software tests and debugging for the same software product independently of each other to collate their respective detected bugs after the tests. On the basis of the proposed model, an optimum release problem for software products is also formulated. The proposed methods are demonstrated by presenting numerical examples to clarify their characteristics.
Design of two‐dimensional IIR digital filters with complex coefficients using spectral transformations
Authors: Sekiguchi Takashi , Takahashi Shin‐Ichi 
Published: July 2007

AbstractThis paper describes a new method for designing two‐dimensional (2‐D) digital filters with complex coefficients having circularly symmetric and fan characteristics by applying variable transformation (spectral variable) to one‐dimensional (1‐D) IIR digital filters with complex coefficients. A 2‐D digital filter with complex coefficients has an advantage in that it can increase greatly the degree of freedom for design, since once it satisfies the design specification partly in the frequency plane, its characteristics can be arbitrary in other regions.Here, a design method is proposed which makes positive use of this property of the complex coefficient digital filters. Since we have introduced the complex coefficient filters into spectral transformation functions and 1‐D prototype filters, there is no constraint on cutoff frequency (radius) accomplishable in circularly symmetric filters. In fan filters even a characteristic that has been difficult to design unless one uses optimization techniques, has become easy to design by means of variable transformation without using any optimization techniques. Moreover, the stability of the resulting IIR filters is guaranteed in simple conditions.