bluuebunny/crossref_metadata_2025_split huggingface dataset:
Synthesis of balanced SC integrator based on bilinear s–z transformation
Authors: Ishikawa Masayuki , Komatsu Yasuaki , Yanagisawa Takeshi 
Published: July 2007

AbstractTo realize an SC integrator which is a representative fundamental section of a switched‐capacitor filter (SCF), via a balanced circuit is effective for high‐frequency applications. to use SC integrators (to which a bilinear s–z transformation is applied) as an s‐region integrator has many advantages in realizing an SCF.This paper proposes a new circuit which realizes an integration characteristic based on bilinear s–z transformation by using a balanced SC circuit. A computer simulation shows that this circuit has a higher frequency range by a factor of about 2, a smaller influence of the capacitor‐error on the integration characteristic, and a smaller common‐mode gain compared with conventional balanced SC integrators with the same scale of hardware. the operation of the proposed circuit was confirmed experimentally by using discrete components.
Synthesis of leapfrog‐type digital filters based on the distributed circuitsimulation
Authors: Iwakura Hiroshi , Tajima Yuji 
Published: July 2007

AbstractIt is known that a leapfrog‐type digital filter obtained by simulating an LC ladder filter terminated with resistors on both ends is a digital filter with a low coefficient sensitivity. In this paper, an analog circuit to be simulated is chosen as a distributed circuit in which the real frequency and the discrete variable correspond rigorously. By means of an equivalent transformation, a leapfrog‐type digital filter is constructed in which no delay free loop is generated. In this method, an LC ladder circuit expressed in terms of the Richard variable p is transformed to the circuit expressed in terms of the LDI variable γ. the digital filter is configured by simulation of the relationship between the node voltage and series branch current in each element of LC ladder circuit. However, if the configuration is based on general distributed LC ladder circuits, a delay‐free loop is always generated and configuration becomes unstable. As a solution to this problem, unit elements are introduced in this paper. It is shown that a realizable signal‐flow graph can always be obtained if stub circuits are used as the original circuit. the basic section necessary to circuit configuration is presented. Finally, as a practical example, the present method is used for design of a fourth‐order bandpass filter. the results are described and compared with those by other configurations.
A consideration for stability problem of leapfrog‐type digital filters and SCFS
Authors: Iwakura Hiroshi 
Published: July 2007

AbstractThis paper describes the stability problem of a low‐sensitivity leapfrog‐type digital filter that simulates the branch voltage‐current of a doubly terminated LC ladder circuit. It has been shown that if the analog transfer function with equal order numerator and denominator polynomials such as a low‐pass, bandpass type circuit whose transmission zero does not exist at infinite frequency, and high‐pass, all‐pass, band‐stop type circuits are synthesized according to this structure, although the transfer function itself is stable, the realized circuit becomes unstable at half the sampling frequency. Hence, the output of this Circuitstill exists even if the input becomes zero.First, it is shown that the Circuitstructure of leapfrog configuration always has an input branch whose transmittance becomes zero at half the sampling frequency. Hence, this structure makes the circuit unstable. When transfer functions of this kind are realized by leapfrog configurations, the stabilization method is proposed by synthesizing the circuit after modifying the transfer function such that the transmission zero exists at infinite frequency. Then this realized circuit is modified again in order to agree with the characteristic of the original transfer function.Finally, design examples are given for the case of a low‐pass switched capacitor filter and a high‐pass digital filter whose transmission zero does not exist at infinite frequency.
Accurate shooting methods for computing periodic solutions of nonlinear systems
Authors: Yamamura Kiyotaka , Horiuchi Kazuo 
Published: July 2007

AbstractThe shooting method is an effective technique for computing periodic solutions of nonlinear systems. In this method, a system of nonlinear integral equations is formulated which has the periodic solution as the solution. Then an iterative method such as Newton's method is applied to the system. However, when high accuracy is required in the numerical solution, a large amount of computation time is necessary because fine steps generally are used for the numerical integration. This paper discusses the application of the Richardson extrapolation to the problem of computing accurate periodic solutions of nonlinear systems. First, it is shown that the error of the numerical solution computed by the shooting method has an asymptotic expansion if the trapezoidal rule is used in the numerical integration. This is easily proved by introducing Keller's theory which is concerned with the finitedifference method. Then it is shown that it is possible to improve the accuracy of the periodic solution significantly by applying the Richardson extrapolation to several numerical solutions with different step sizes. It is also proved that Newton's method converges quadratically if the periodic solution is isolated and the step sizes are sufficiently small. Some numerical examples are given to show the effectiveness of the approach presented here.
Improvement of convergence characteristics pole‐controlled IIR echo canceller
Authors: Takumi Ichi , Hata Masayasu 
Published: July 2007

AbstractThe authors have proposed an echo canceller using a parallel connection of independently controllable second‐order IIR adaptive digital filters (ADF5). In this canceller, each second‐order IIR ADF can be controlled independently. After the pole of the ADF has been arranged approximately ac cording to the transfer characteristic of an echo path, the pole is moved toward the pole of the echo path by a precision control. This precision control does not work some times depending on the position of the approximate arrangement.This paper describes problems of the forementioned precision control of the pole and their causes. This paper also describes an equal‐echo‐return loss enhancement (ERLE) contour which is obtained from the relationship between the second‐order IIR ADF and the pole of the echo path. A new method of determining an additional pole (for the precision control) by introducing the equal‐ ERLE contour is proposed. the results of a simulation of the precision control of the pole using the proposed method are shown. It is seen from these results that an echo canceller using the proposed method can obtain more than twice the conversion speed of the previously proposed method.
A high‐frequency compensation of sallen‐key circuits based on pole sensitivity
Authors: Okine Mitsuo , Fujii Nobuo 
Published: July 2007

AbstractThe Sallen‐Key circuit is one of the most practical circuits for active RC filter. As is well known, it is composed of the amplifier with a finite gain, for which an operational amplifier has been employed from the viewpoint of the cost and the easiness of the design. On the other hand, due to the finite gain‐bandwidth (GB) product of the operational amplifier, the cutoff frequency of the realized filter is restricted to below 10 kHz. This paper proposes a design method where a passive element is added to the Sallen‐Key circuit. By this scheme, the effect of the finite GB product is compensated, and a high‐frequency circuit is realized. the pole‐sensitivity of the transfer function with respect to GB product is also reduced. By employing the proposed method, not only the effect of the first‐order pole on the transfer characteristic due to the change of GB product, but also the effect of the second‐ order pole can be minimized. Finally, as a design example, a bandpass filter with fO = 100 kHz and Q = 1 is constructed. Its operation was verified by experiment.
An adaptive filter based on singular‐value decomposition of input signal correlation matrix
Authors: Kikuchi Yoshihiro , Saito Tsuneo 
Published: July 2007

AbstractAlthough various algorithms have been proposed for updating the filter coefficients of adaptive filters, problems still remain concerning their convergence speed and the accuracy of filter coefficients after the convergence. to solve these problems, this paper proposes an adaptive algorithm in which filter coefficients are obtained by solving a normal equation (for the optimized filter coefficients or the Wiener‐Hopf equation using eigenvalues of the input‐signal correlation matrix. If there are small eigenvalues in the input‐signal correlation matrix, they are assumed to be zero so that the filter coefficients are calculated by using a generalized inverse matrix. This prevents the increase of errors which occurs when the input is a correlative smooth signal. the proposed method matches nonsteady‐state signals, since the eigenvalues are successively calculated using the rank‐one‐modification method.The characteristics of the proposed adaptive filter were investigated, and a computer simulation of the filter as a noise canceller was carried out. the results show that the characteristics of the filter are superior to those of filters using conventional algorithms, although the amount of computation increases considerably.
Untraceable electronic funds transfer systems
Authors: Tanaka Yoshiaki , Akiyama Minoru , Uchida Takanori 
Published: July 2007

AbstractThe transfer of funds by electrical communication contains several problems. the electronic record, for example, can easily be copied, and it may be possible to make more than one payment on the same electronic check. From the viewpoint of privacy protection, it is desirable that the flow of funds cannot be traced. This paper proposes a method of money transfer and checking to solve those problems. Especially, a tally check system is proposed for the check system whereby privacy is protected. In the proposed method, before sending money the sender receives the tally constructed by the receiver, and asks the bank to issue an electronic check based on that tally. When the receiver wants money, he presents the electronic check to the bank together with the other tally which is secretly retained. Since the tally is used as proof for the correct receiver, it is impossible for the bank to trace the flow of funds. It is impossible to cash a check more than once since the bank retains the presented tally as the receipt.
Error correction for shaft encoders using M‐sequences
Authors: Nomura Eiji , Yamaguchi Kazuhiko , Imai Hideki 
Published: July 2007

AbstractA method known to be useful in detecting the rotation angle of a device such as motor, is to attach an encoder with M‐sequence to the shaft and to read the angle of the encoder by a sensor. However, in the past, the reading error as the sensor has not been considered in such a shaft encoder using the M‐sequence. This paper proposes a method for error detection and correction. When a reading error is produced in the symbol read by the sensor, it is detected and corrected based on the linear recurrence relation of the M‐sequence. For some given requirements for the measurement accuracy, the number of symbols needed for single error detection and correction is investigated.
Real and artificial speech signals for objective quality evaluation of speech coding systems
Authors: Itoh Kenzo , Kitawaki Nobuhiko 
Published: July 2007

AbstractThis paper discusses the test signals to be employed in the objective evaluation of the speech coding system. the dependency of the objective evaluation measure on the speaker is examined. the feature parameters of the speech, which are largely responsible for the dependency, are discussed. the number of speakers and the length of the speech sample required in the stable objective evaluation are examined. As the result, it is concluded that one should be careful about the dependency on the speaker when the real speech signal is used as the test signal. It is desirable to use a speech sample of longer than 4 to 5 s uttered by some 10 speakers. Then the test signal other than the real speech is discussed. Artificial speech, which approximates the distribution of the speech signal on the frequency domain, is discussed. A new artificial speech (ASVQ) is proposed based on the vector quantization and the speech synthesis technique. to verify the usefulness of ASVQ, examinations are made for a typical speed coding system using distortion measures in the time and frequency domains. As a result, it is seen that ASVQ represents the features of the speech more efficiently than other artificial speeches which have been proposed for various purposes. It is shown that by employing ASVQ as the test signal, an objective evaluation very close to the value by the real speech can be obtained.
Design of window function using spline functions
Authors: Toraichi Kazuo , Mori Ryoichi , Kamada Masaru 
Published: July 2007

AbstractIn our previous paper, it was shown that the rectangular, triangular and Parzen windows can be generalized as a B‐spline function according to its order. That series of window functions has a merit in that a fast fall‐off of the sidelobe is realized by increasing the parameter, i.e., the order, while it has a demerit in that the bandwidth of the mainlobe is increased. This paper aims at the improvement of this point, and proposes a design method whereby the window function is represented by a linear combination of B‐spline functions. As an example of the window function obtained by the proposed method, a window function is presented which has the same mainlobe bandwidth and highest sidelobe level as Hanning or Blackman window, and still has a faster fall‐off. Almost the same quality of approximation as the Hanning or Blackman window is also demonstrated.
Evaluation for quality of speech synthesized by a method based on ARMA model
Authors: Wang Gang , Miki Nobuhiro , Nagai Nobuo 
Published: July 2007

AbstractThis paper considers the use of the ARMA model in speech synthesis and discusses the quality improvement and problems in synthesized speech. First, the construction of the analysis filter is considered which is used to determine the residual signal in the speech synthesis by the ARMA model. It is pointed out that special considerations must be given to the MA part. Then the qualities of the speech synthesized by ARMA coefficients and the speech synthesized by AR coefficients are compared. the quality of speech was evaluated based on the preference measure in the listening test (Sheffé's paired comparison). It is shown that the quality of the synthesized speech is better when ARMA coefficients are employed. It is also shown that the quality of the synthesized speech in the AR model is not affected greatly by the method of analysis. Finally, a new measure for the spectral distortion is proposed as a measure for objective evaluation of the quality of the synthesized speech. the measure considers the properties of the MA part of the ARMA model and gives a result which corresponds well with the result of evaluation by the listening test.
Variable switched‐capacitor bandpass, band‐elimination filters and all‐pass networks with low sensitivities
Authors: Kato Kyoko , Kubota Katsuhiro , Takebe Tsuyoshi 
Published: July 2007

AbstractThis paper proposes a design method for the second‐order switched‐capacitor (SC) band‐pass filter (BPF) with high‐Q and low sensitivities, whose center angular frequency ω0 and bandwidth can be varied with a single capacitor respectively. It is shown also that a band‐elimination filter and an all‐pass network with similar variable characteristics can be realized by the same circuit topology. the proposed method is based on the low‐pass to bandpass frequency transformation realized by replacing the delay element in a first‐order low‐pass filter (LPF), whose bandwidth is variable by a single capacitor, by a unit function block with variable phase by a single element. to realize a filter with high‐Q and low sensitivities, a unit function with a fast‐phase rotation around ω0 must be used.Such a unit function cannot be realized with the digital circuit because the circuit has a delay‐free loop. However, the delay‐free loop can be permitted in the SC network, and therefore the desired unit function can be realized by the SC network. the proposed method takes this SC advantage; ω0 is variable by a single capacitor contained in the unit function block. the SC network is realized using six operational amplifiers for which no dynamic range adjustment is required. After discussion the sensitivities in the proposed network, design examples of all three kinds of networks with Q = 36 and operating at clock frequency of 25 kHz are shown with experimental results.
A full‐duplex analog speech scrambler using the FFT technique and its performance
Authors: Matsunaga Akira , Ohkawa Michihisa , Koga Keiichiro , Sakurai Keiichi 
Published: July 2007

AbstractThe FFT scrambling method based on the manipulations of the Fast Fourier Transform (FFT) coefficients of speech is one of the analog speech scramblers which realize high‐level security and excellent descrambled speech quality. It is not clear, however, how the descrambled speech quality and syn‐chrnoization performance are affected by transmission impairments. This paper analyzes the effects of transmission impairments on the descrambled speech quality by computer simulations and provides a guide for the system design. Then practical methods to cope with the effect of transmission impairments are proposed.Based on the results of the simulations, experimental scramblers are developed using general‐purpose DSP's (Digital Signal Processors). The system configuration and the result of the performance evaluation tests are shown. The experimental scramblers are equipped with an automatic equalizer for group delay and amplitude distortions, as well as the automatic frequency control (AFC). An echo canceller is also provided so that the full‐duplex operation is achieved using two‐wire lines. As a result of the test under the simulated environments, it is proved that the descrambled speech quality is satisfactory and the proposed method is effective in practice.
A new class of 2‐D digital filters composed of all‐pass subfilters
Authors: Toyoshima Hisamichi , Ikehara Masaaki , Takahashi Shin‐Ichi 
Published: July 2007

AbstractThis paper proposes a new construction method for 2‐D IIR digital filters with all‐pass subfilters as components. Recently, numerous studies have been made to construct digital filters by combining all‐pass characteristics. This paper is based on the construction by a parallel connection of an all‐pass subfilter and delay element, which can realize approximately the linear phase characteristic. First, the basic design for 1‐D digital filters as well as its properties are described. Then the theory for the 1‐D case is extended to the 2‐D case, and a new construction method for 2‐D digital filters with low‐pass characteristics with circular symmetry is shown. The proposed method features the following. The phase characteristic is approximately linear in the passband. The filter can be realized with a smaller number of multipliers than in the traditional method. The coefficient sensitivity is low.
Construction and performance evaluation of fermat codes
Authors: Yamanishi Kenji 
Published: July 2007

AbstractThe algebraic geometrical codes proposed first by Goppa are a family of linear block codes which are generated from algebraic curves. Among those, this paper considers especially Fermat codes which are generated on Fermat curves. The method for constructing Fermat codes is presented and its performance is evaluated for the finite‐length code. It is shown as a result that Fermat codes are promising as code with good parameters, which can systematically be constructed. In this paper, as the first step, several structures of Fermat curves are presented which are needed in the construction of Fermat codes. Based on these structures, the general expression forms for the generator matrices of Fermat codes are derived. Then the codes generated based on Fermat codes in the form of subfield subcodes or concatenated codes are considered. The precise parameter values are evaluated for the codes on F2m(m = 1, 2, 4, 6, 8) and they are compared with the conventional algebraic codes.
Variational analysis of new shape face shear mode quartz crystal resonator
Authors: Kawashima Hirofumi , Matsuyama Masaru , Nakazato Mitsuhiro 
Published: July 2007

AbstractThis paper describes a variational analysis, the frequency temperature characteristics and the electrical equivalent circuit constants of a face shear mode quartz crystal resonator with two supporting portions at the ends formed by an etching method. The object of this paper is to propose a new shape face shear mode quartz crystal resonator and to clarify its frequency temperature characteristics and its electrical equivalent circuit constants in the calculation and in the experiments. A vibration analysis is performed according to the following procedure.First, a frequency equation is derived by performing a vibration analysis of a face shear mode resonator being incorporated with a vibrational portion and supporting portions with an energy method. From the obtained frequency equation, a relationship of frequency temperature coefficients vs. a cut angle or a mass ratio is clarified.Second, some of the calculated results are compared with the experimental value. As a result, the calculated and experimental results show good agreement.Finally, various constants of an equivalent circuit for quartz crystal resonators are examined. It is shown that a face shear mode quartz crystal resonator with low series resistance and a high quality factor in the frequency range of 1.5 to 4.2 MHz is obtained, even if supporting portions are added and connected to the vibrational portion.
Shiftcompaction—quasi‐two‐dimensional compaction method for symbolic layout
Authors: Sakamoto Mamoru , Onodera Hidetoshi , Tamaru Keikichi 
Published: July 2007

AbstractA quasi‐two‐dimensional method called shiftcompaction has been developed as a compaction method for symbolic layout. Shiftcompaction takes a one‐dimensional compaction result as an input and tries to reduce the layout width by cutting off critical paths by moving elements on these paths perpendicularly to the compaction direction.In other words, it is a method for realizing process efficiency and effective layout compaction by restricting the objectives to be investigated for two‐dimensional translation to the layout elements that determine the current layout width. Furthermore, with a combination of a jog insertion operation, it can achieve denser layout compaction.Since those layout elements to be translated in the anti‐compaction direction are determined based on analysis of placement constraints in the anti‐compaction direction, it is possible in shiftcompaction to control the layout width in the anti‐compaction direction. As the results of compaction experiments, when, for example, a layout with 57 blocks and 110 wires was compacted, a 19 percent smaller layout area than a one‐dimensional compaction was obtained. This result is 6 percent smaller than that due to combination of one‐dimensional compaction and jog insertion. This verifies the effectiveness of the compaction method presented here.