bluuebunny/crossref_metadata_2025 huggingface dataset:
Insecticidal activity of essential oil from Chamaecyparis formosensis Matsum
Authors: Kuo Pei-Min , Chu Fang-Hua , Chang Shang-Tzen , Hsiao Wen-Feng , Wang Sheng-Yang
Published: July 2007

Chamaecyparis formosensis is an endemic precious tree species grown in Taiwan. Based on the results obtained in our current study, the essential oil of C. formosensis wood possesses significant insecticidal activities. In an assay to assess mosquito larvicidal properties, the essential oil completely killed the larvae of Aedes aegypti and Aedes albopictus at a dosage of 100 μg ml-1 in water. Concentrations of the essential oil at a dosage of 0.16 mg cm-3 in air resulted in 100% mortality of silverfish after 2 h. It was found that the extractives of C. formosensis possess multiple bioactivities against insects. There is great potential for the development of insecticides for domestic application to kill mosquito larvae and silverfish based on this raw material.
Gaur (Bos gaurus) and Banteng (B. javanicus) in the lowland forest mosaic of Xe Pian Protected Area, Lao PDR: abundance, habitat use, and conservation
Authors: Steinmetz Robert
Published: July 2007

Gaur (Bos gaurus) and banteng (Bos javanicus) populations throughout South-east
Asia have declined severely because of hunting and habitat fragmentation. Important
remnant populations persist in Xe Pian national protected area in southern
Lao P.D.R., where sign-based surveys were carried out between 1996 and
1998 to determine their distribution, abundance, and patterns of habitat use.
Xe Pian is comprised of a largely intact lowland mosaic of semi-evergreen,
mixed deciduous, and dry dipterocarp forest types. Gaur used a broader
diversity of these habitat types than banteng, attaining moderate densities in
homogeneous semi-evergreen forest as well as expanses of deciduous dipterocarp
and mixed deciduous forests. Mixed deciduous forest was the least abundant
forest type but was commonly used by gaur. Banteng showed a strong affiliation
with drier and more open habitats, especially dry dipterocarp forest, despite
increased vulnerability to hunting in these areas in the past. Banteng were not
found within large expanses of semi-evergreen forest. Their distribution within
Xe Pian was therefore more restricted than gaur, though they were relatively
more numerous within two isolated corners of the protected area. Signs of
calves and juveniles indicated that both species retained breeding populations
in Xe Pian. Remaining herds were small – composed of two to five individuals
– but bamboo understories in semi-evergreen forest were a food source that
attracted larger congregations of gaur in the rainy season. The banteng
population in Xe Pian is globally significant for conservation, while that of
gaur is nationally significant. The existence of extensive high quality habitat
and on-going collaboration of local people lends hope that Xe Pian's wild cattle
will increase, given protection from hunting.
Biochemotherapy of metastatic melanoma in patients with or without recently diagnosed brain metastases
Authors: Majer Martin , Jensen Randy L. , Shrieve Dennis C. , Watson Gordon A. , Wang Michael , Leachman Sancy A. , Boucher Kenneth M. , Samlowski Wolfram E.
Published: July 2007

BACKGROUND.

Brain metastases are an alarming complication of advanced melanoma, frequently contributing to patient demise. The authors performed a retrospective analysis to determine whether the treatment of metastatic melanoma with biochemotherapy would result in similar outcomes if brain metastases were first controlled with aggressive, central nervous system (CNS)‐directed treatment.

METHODS.

Seventy melanoma patients were treated with biochemotherapy for metastatic melanoma between 1999 and 2005. Of these, 20 patients had recently diagnosed brain metastases, whereas 50 did not. Brain metastases (if present) were treated with stereotactic radiosurgery ≥28 days prior to systemic therapy. All patients were treated with biochemotherapy consisting of either dacarbazine or temozolomide in combination with a 96‐hour continuous intravenous infusion of interleukin‐2 and subcutaneous interferon‐α‐2B. The primary endpoint was survival from the time of the initial diagnosis of metastatic disease.

RESULTS.

Median survival from the time of the diagnosis of metastatic melanoma was 15.8 months for patients with brain metastases and 11.1 months for those without CNS involvement (P = .26 by the log‐rank test; P = .075 by the Gehan Wilcoxon test). Dacarbazine‐based and temozolomide‐based regimens appeared similar with regard to their effect on overall survival and CNS disease progression. A plateau in further brain recurrences was observed in patients who survived for > 20 months.

CONCLUSIONS.

Data from the current study suggest that the outcome of biochemotherapy is comparable in patients with and those without brain metastases, if brain metastases are controlled with multidisciplinary treatment. Prolonged survival can be achieved in approximately 15% of patients, regardless of whether or not brain metastases are present. Cancer 2007. © 2007 American Cancer Society.
Low‐Temperature Growth of p‐Type ZnO Thin Films via Plasma‐Assisted MOCVD
Authors: Lu Y.‐F. , Ye Z.‐Z. , Zeng Y.‐J. , He H.‐P. , Zhu L.‐P. , Zhao B.‐H.
Published: July 2007

N‐doped p‐type ZnO films are grown on glass substrates by plasma‐assisted metal–organic CVD at a low temperature of 250 °C. Scanning electron microscopy, X‐ray diffraction, and Raman scattering spectroscopy are carried out to investigate the crystalline properties of the ZnO films. Results show a high degree of (0002) orientation and a little tensile stress in the films. Hall analyses show homogeneous p‐type conduction. Room‐temperature photoluminescence spectra display strong ultraviolet emission with a clear acceptor fingerprint.
Ion‐Exchange Plasma Membranes for Fuel Cells on a Micrometer Scale
Authors: Roualdès S. , Schieda M. , Durivault L. , Guesmi I. , Gérardin E. , Durand J.
Published: July 2007

Recent advances in miniaturization technology make polymer electrolyte membrane fuel cells very attractive as power sources for portable devices. Ion‐exchange membranes for microscale fuel cells are synthesized by plasma polymerization (using a precursor containing ion‐exchange groups) and intensively characterized. Ion‐exchange plasma membranes are thin, amorphous, and dense materials with no defects. Spectroscopic analyses reveal a polymer‐type matrix containing a rather high concentration of ion‐exchange groups. Under the best synthesis conditions, membranes show a satisfying ionic conduction level and a high compatibility with other active layers of fuel cells, making them suitable for insertion in such power‐supply devices.
Synthesis of balanced SC integrator based on bilinear s–z transformation
Authors: Ishikawa Masayuki , Komatsu Yasuaki , Yanagisawa Takeshi
Published: July 2007

To 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

It 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

This 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

The 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

The 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

The 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

Although 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

The 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

A 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

This 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

In 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

This 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

This 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

The 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

This 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

The 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

This 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

A 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.
Spectral line estimation using complex coefficient prefilters
Authors: Shimamura Tetsuya , Takahashi Shin‐Ichi , Sakaguchi Naoya
Published: July 2007

A spectral line estimation method is proposed which improves the resolution of the two nearby sinusoidal frequencies. In contrast to the conventional method in which the decimation operation is applied to the signal passing through the real coefficient bandpass prefilter, the present method carries out the process by means of the complex coefficient prefilter. By means of this technique, numerous problems to be resolved under the condition that relatively few data are given under a low SN ratio, such as the problem of initial phase dependence and the problem of accurate spectral estimation at a low‐ or high‐frequency region, can be overcome.

This paper presents a method in which the decimated signals by a decimation operation are used effectively. The method particularly improves the estimation capability in the case with fewer data. Finally, by means of a simulation, the effectiveness of these methods is proven. It is shown that the estimation accuracy is improved significantly as long as the amount of operations is similar to that for the conventional method using the real coefficient prefilter.
Computationally efficient determination of a polynomial discriminate function using the AIC and its systolic array implementation
Authors: Iiguni Youji , Sakai Hideaki , Tokumaru Hidekatsu
Published: July 2007

This paper presents a computationally efficient algorithm for selecting the terms of the polynomial discriminate function (PDF) by means of the Akaike information criterion (AIC). The Levinson‐type algorithms is introduced for a computationally efficient determination of the PDF and a simple computation of the AIC. Our method is applied to seven segment and vowel recognition to show its usefulness. In an actual application, a parallel information processing is indispensable to reduction of computation time. Thus, systolic array for computing the PDF that is suitable for VLSI implementation is shown.
Approximate realization of identity mappings by three‐layer neural networks
Authors: Funahashi Ken‐Ichi
Published: July 2007

Recently, applications of neural networks to information compression, e.g., image data compression, have been studied. This paper discusses these problems from a viewpoint of the approximate realization of identity mappings by the use of three‐layer neural networks.

The relationship between statistical properties of data, hidden units number and the infimum of approximation error also is clarified theoretically by the study of the relation with principal component analysis in the case in which the hidden layer generally has nonlinear units. As a result, it is proved that the capability of information compression by three‐layer networks does not exceed the capability of the K‐L transformation method. Moreover, the phenomenon of generalization in this case is explained theoretically.
Trellis coded modulation for additive impulsive noise
Authors: Ogiwara Haruo , Ootaki Ryuichi
Published: July 2007

This paper discusses the situation where the trellis coded modulation (TCM) is applied to the digital subscriber loop (DSL) using a balanced pair cable, and examines the effectiveness. The usual error‐correcting code cannot be applied successfully to the balanced pair since the loss increases with the increase of the transmission bandwidth. When TCM is applied, on the other hand, the transmission performance for the Gaussian noise can be improved without increasing the transmission bandwidth. The major factor deteriorating the performance in DSL, however, is the impulsive noise. From such a viewpoint, this paper examined the effectiveness of TCM for the impulsive noise.

First, the probability density of the impulsive noise is derived from the measurement of the exiting DSL, and the channel capacity is calculated. It is shown that it is necessary and sufficient to double the transmitting symbols and to assign the produced redundancy to the error‐correction. This is the same as the known result for the Gaussian noise, but the coding gain by the ideal coding is 18 dB, which is larger than in the case of the Gaussian noise. The coding gain of 9 dB is obtained by the simulation using the 8‐state convolutional code, which is larger than 4 dB in the case of the Gaussian noise. The reason for this improvement is discussed from the viewpoint of the probability distribution of the noise and the signal mapping in the multidimensional signal space.
Adaptive arrays using a complex learning algorithm
Authors: Kobayashi Akira , Yahagi Takashi
Published: July 2007

The demand for the mobile communications is increasing every year, and TV and automobile telephone by satellite communication are expected to find wider applications. If individuals are to carry communication equipment, then the equipment must be small size, lightweight and highly efficient. To realize antennas which must also be of small size, lightweight and highly efficient. For such equipment it is desirable to employ the adaptive array with phase control.

Conventionally, the LMS algorithm has been applied frequently in the design of the adaptive array. However, the authors believe that the learning algorithm is better suited to the adaptive array because of its higher convergence speed and the clear specification of the convergence condition.

This paper extends the real‐type learning algorithm to the complex‐type learning algorithm. The convergence condition for the algorithm is derived, and the result of the application of the method to the adaptive array is shown. A method of deriving the reference wave also is presented which is less affected by the input signal variation and is suited to the adaptive array with only the phase control. Using the proposed complex learning algorithm and the new reference wave, the phase‐controlled adaptive array is designed. A satisfactory result is obtained by simulation.
A realization of interpolator with the coefficients of the power of two
Authors: Todaka Chiaki , Kambayashi Noriyoshi
Published: July 2007

This paper is aimed at the reduction of the hardware complexity and the computational complexity in the interpolation processing. A construction and design method for the interpolator is shown where only the coefficients of the power of two are employed. The interpolator is composed of the cascade connection of a passband compensatory filter, a zero‐interpolator, and a stopband forming filter. Since no multiplier is used in the proposed construction, zero coefficient sensitivity as well as a high‐speed operation are realized. FIR or IIR construction is used for the passband compensatory filter, and the stopband forming filter is constructed by a cascade connection of RRS.

The design procedure for the interpolation is as follows. The linear programming is applied first to design the stopband forming filter. Then the passband compensatory filter is designed by the indirect exhaustive method or the least‐square method. It is shown through design examples that the proposed interpolator is useful in reducing the hardware complexity and the computational complexity.
Orthogonal periodic sequences with two complex numbers derived from M‐sequences
Authors: Kirimoto Tetsuo , Oh‐Hashi Yoshimasa , Hotta Takashi
Published: July 2007

The periodic sequence for which the sidelobe of the autocorrelation function is zero is called an orthogonal sequence. The orthogonal sequence is applied to various problems such as synchronization of communication and radar ranging. As a general method to generate the orthogonal sequence, the method to form the discrete‐Fourier transform of the periodic sequence with a constant amplitude is known.

This paper proposes a method of generating the orthogonal sequence with an element of the sequence being composed of two complex numbers (orthogonal periodic sequence with two complex numbers). The method does not use the discrete‐Fourier transform, but maps the elements of the M‐sequence on Galois field GF(2) to complex numbers. Then the properties of the orthogonal sequence are discussed from the geometrical viewpoint and the range for which the orthogonal periodic sequence with two complex numbers exists is indicated.
A study on coding of fractal curves
Authors: Katsuno Shin'Ichi , Yamasaki Issei
Published: July 2007

This paper describes the feature of codings for fractal curves. Here the two basic coding methods, the chain coding and the polygonal approximation, are examined. The ratio of the number of chain codes to the number of line segments for fractal curves is derived as a function of the triple quantities: the maximum allowable error; the line length of polygonal approximation; and the coefficient which is determined by the method of fractal curve generation. The ratio of the maximum allowable error to the line about the polygonal approximation of the artificial fractal curve can be derived from its fractal dimension.
A synthesis of active RC multipassband filters
Authors: Watanabe Hiromichi , Kikuchi Hisakazu
Published: July 2007

This paper presents a synthesis method for active RC filters with multiple pass‐bands. The synthesis procedure is based on a signal‐flow graph representation and frequency transformation applied to a passive LCR filter. It should be noticed that an active RC filter with multiple passbands is amenable to the finite gainband product effects of an operational amplifier. This is really true if adjacent passbands are close. Hence an effective technique is developed for reducing the finite GB product effects. An experimental result concerning the two‐passband filter derived from a third‐order Chebyshev filter also is included.
Realization of resistively matched three‐ports and the suppression of multiple reflections traveling on branched transmission lines
Authors: Sakagami Iwata
Published: July 2007

In the mutual interconnects between the printed transmission lines containing highspeed logic devices, impedance mismatch causes multiple reflections of the pulse waveform so that false triggerings of logic device occur. This paper describes suppression of multiple reflections on a three‐port single‐branched transmission line which divides a signal into two directions.

First, a resistively matched three‐port is proposed in which all the ports are matched to arbitrary loads. Its realization by the scattering matrix is shown. If negative branching resistors are included, this circuit can always be realized, although a negative resistor is not easily conceived. Hence, the range of the load resistance is found which is realizable with a nonnegative branching resistor. This range is shown graphically.

Next, three transmission lines with arbitrary characteristic impedances are connected to the resistively matched three‐port. This is used as a three‐port, single‐branched transmission line. This circuit supports an ideal condition in which a signal is sent out to the output side without reduction of the amplitude level and without causing any multiple reflection for an incident voltage wave. However, this ideal condition has two cases: one requiring a negative branching resistor, and another requiring an infinite characteristic impedance. Neither can be realized easily. Hence, realizable element values closest to the ideal condition are selected for which the multiple reflection is studied. In comparison with the case without a resistive matched three‐port, the effectiveness of suppression of the multiple reflection is superior if the latter is used.
Schematic description language for a hardware top‐down design approach
Authors: Kobayashi Kazuo , Wakabayashi Haruo , Wakimura Yoshiaki
Published: July 2007

This paper presents a schematic language for describing the functional specification of hardware. The language provides a new computer input medium using a top‐down design essential to the systematic design of large‐scale hardware. This language has a special description model that uses a signal based on a new concept of connections between function blocks, called a virtual interface signal. This signal allows a designer to refer to it unrestrictedly among a hierarchy of function blocks. Therefore, the design can describe both an outline of the connections at an early design stage and the detailed connections at the final design stage based on the same description rules. Moreover, the designer can specify the hardware function by means of both a function block and a state transition diagram.

Some preliminary experiments show that the language herein requires only about a quarter of the description man‐hours as that of a conventional character‐based description language. Also, using two types of design diagrams reduces the description man‐hours by about 40 percent over the function block diagram only.
Design of multiple‐valued module array using signed‐digit arithmetic
Authors: Kameyana Michitaka , Higuchi Tatsuo , Nomura Masahiro
Published: July 2007

A new design methodology for arithmetic VLSI systems is proposed using a radix‐4 minimum‐redundant signed‐digit (SD) arithmetic module. The module is composed of an adder, a sign inverter and a partial product generator. Any arithmetic systems based on addition, subtraction and multiplication can be realized by appropriately specifying interconnections between the modules. This module array is very useful for short design turnaround time. Moreover, the operating speed of the proposed module array systems is comparable to that of the fastest binary systems.
Identifiability of hidden markov processes and their minimum degrees of freedom
Authors: Ito Hisashi , Amari Shun‐Ichi , Kobayashi Kingo
Published: July 2007

There exists a class of stochastic processes based on a finite‐state discrete‐time Markov chain, where the states are not directly observable and only the output symbols generated by the states are observable. If different states produce the same output symbol, the process is called a hidden Markov process. It is very important both in theory and applications. It is known that two Markov processes with different transition matrices are equivalent hidden Markov processes. The problem has been considered interesting concerning by what structures such a situation can arise.

This paper gives a complete solution to the identification problem of the hidden Markov process, which has long been proposed and is still unsolved. At the same time, a structure of this kind of process, which has been unknown, is revealed. An n algebraic technique is used in the discussion, where new concepts called Δ‐subspace and Δ‐cy‐clic subspace are introduced into the framework of subspace and cyclic subspace in the usual linear algebra.

In this paper, a Δ‐subspace is actually constructed which gives the effective minimum degree of the hidden Markov process. It is shown also that two stochastic processes are equivalent if and only if there exists an isomorphism between the Δ‐sub‐spaces.
SPRODICE: A diagnosis support program for the interpretation of brain CT images
Authors: Yan Zhuang‐Zhi , Eiho Shigeru , Tanaka Hiroshi
Published: July 2007

This paper describes the knowledge base system SPRODICE, which is constructed to support the interpretation and diagnosis of brain CT images. The knowledge base of the proposed system has the knowledge about the internal structure of the brain, and diseases are represented by a hierarchical semantic network, and the sites of common incidence for each disease as well as the finding patterns based on the intuition and experience of the doctor are described by the rules.

The system positively introduces the doctor's decision in the course of the diagnosis so that the diagnosis is not left to the system, but proceeds by the cooperation of the doctor and the system. This approach originated from the idea that the final responsibility of the result of diagnosis lies with the doctor.

It is made possible also to complement the incompleteness and insufficiency of the knowledge, which is always a problem in the diagnosis‐support system, by the common sense and the experience of the doctor. The system is implemented on a personal computer using Prolog, and can support approximately 80 kinds of interpretation and diagnosis of brain diseases. As a result of experimental diagnosis of 107 pathological cases, it is seen that the accuracy of diagnosis by SPRODICE is 82.2 percent when used by a non‐specialist, and 83.4 percent when used by a specialist.
Designing method for ARMA four‐line lattice filter with sliding rectangular window
Authors: Haseyama Miki , Nagai Nobuo , Miki Nobuhiro
Published: July 2007

This paper derives a design method for an ARMA 4‐line lattice filter using a sliding rectangular window. The adaptive ARMA 4‐line lattice filter already proposed uses a forgetting factor, which is one of the weighting functions to estimate coefficients of a time‐varying system in which system coefficients vary with sufficient smoothness. Therefore, the effect of past observed signal over the estimated coefficients decreases exponentially.

The filter presented here is realized using a rectangular window because the concern is over the effects of past observation signal rather than window length. Using this filter, an input signal is estimated when designing; furthermore, a system can be identified in which an arbitrary section in the frequency domain is weighted.

Thus, by not only analyzing voice signal which is considered a model whose input signal is unknown and EEG data but also by weighting the frequency domain, for example, the holmant in the low‐frequency domain can be estimated in low degree with very high precision and a specific wave (such as a α activities) in EEG data can also be detected. Moreover, in this paper the algorithm is verified by model experiments.