Showing papers in "Electronics and Communications in Japan Part Iii-fundamental Electronic Science in 1989"

Secret sharing scheme realizing general access structure

Abstract: As a method of sharing a secret, e.g., a secret key, Shamir's (k, n) threshold method is well known. However, Shamir's method has a problem in that general access structures cannot be realized. This paper shows that by providing the trustees with several information data concerning the distributed information of the (k, n) threshold method, any access structure can be realized. the update with the change of the secret trustees and the relation to the threshold graph are also discussed.

A generalized sampling theorem

Abstract: Using the theory of pseudo-biorthogonal base and the notion of reproducing kernel, we derive a very broad generalized sampling theorem with real pulse. In addition, it includes the traditional sampling theorem with ideal pulse as its special case. It also includes all the sampling theorems in the cases of bandpass-type band-limited signal space for nonuniformly spaced sampling points, for many variables, and for the case in which the notion of frequency is extended from Fourier transform to general integral transform. This generalized sampling theorem is effective also for so-called undersampling whereby there are too few sampling points compared with the dimension of signal space, and for so-called oversampling due to too many sampling points. Moreover, it is effective in the case where both occur; that is, in the case where it is undersampling from the standpoint of signal space while it is oversampling in a space of restored signal. For undersampling, the generalized sampling theorem provides the best approximation for each original signal.

A block placement procedure using a force model

Abstract: The paper treats the problem of automatic placement of building blocks of different sizes. This problem arises frequently in automating the layout design of LSIs. In this paper, a method is described for finding a global placement taking into account the connectivity between blocks as well as dimensions of blocks and placing regions. A method is proposed to establish a force model such that attractive forces caused by the connectivity between blocks and repulsive forces caused by the block overlaps are exerted on the blocks, and to derive a placement result as a system equilibrium state. the placement procedure is as follows. We start with the initial placement obtained only by applying attractive forces caused by connectivity, and then add gradually repulsive forces which are calculated based on the actual overlapping area between blocks. As a result, block overlap is removed and thus we can obtain a global placement considering local shape fitness while taking connectivity between blocks in a global way. In this paper, we explain first an outline of the proposed algorithm and then describe details of the methods for realizing it, e.g., functional forms of repulsive force against overlapping area, increasing ratio of repulsive force, and calculation method for the block equilibrium position and block orientation. Finally, the experimental results using examples with 17 and 33 blocks to demonstrate the effectiveness of the proposed algorithm are explained.

Propagation characteristics of plate waves in a Z-cut X-propagation LiTaO3 thin plate

Abstract: This paper describes the numerical analysis and experimental verification of the behavior of plate waves in a Z-cut X-propagation LiTaO3 thin plate. With the product of the frequency and the substrate thickness as a parameter, the phase velocity, mechanical displacement and electromechanical coupling coefficient are obtained. the Lamb waves of zeroth-order antisymmetric mode (AO) and symmetric mode (SO) are coupled more tightly than the Rayleigh mode. the SO mode SH wave is a tightly coupled mode comparable to the piezoelectric plate, and is of low dispersion. the results of an experimental verification in an LiTaO3 of 100 μ thick agreed well with the theoretical data in the frequency range below 50 MHz. These results are effective in designing the plate wave devices.

High‐speed decoding of BCH codes using a new error‐evaluation algorithm

Abstract: A 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.

Notes on q‐ary expanded Reed‐Solomon codes over GF(qm)

Abstract: Error-correcting techniques are important in digital communications, for example, satellite communications, mobile communications, and digital audio systems. In digital communications, block codes such as Reed-Solomon (RS) codes (recommended for compact disc (CD) and digital audio tape (DAT)) are used. This paper treats q-ary expanded RS codes over GF(qm) as two-dimensional codes over GF(q) and gives a new interpretation to q-ary codes by utilizing the concept of superimposed concatenated codes. Using this new interpretation a lower bound on the minimum distance of q-ary expanded RS codes over GF(qm) is derived and a new decoding algorithm is clarified.

A chaotic circuit family including one diode

Abstract: This paper considers the chaotic phenomena in a family of three-dimensional autonomous circuits composed of R, L, C, -R (linear negative resistor) and one diode. First, a canonical equation is proposed which describes such a family of circuits. It is a piecewise-linear equation containing one large parameter α. When a is sufficiently large, an idealization applies where the diode is regarded as a switch. This idealiza tion is valid theoretically. the equation is simplified to a system where the two- and three-dimensional linear equations alternate. the one-dimensional Poincare mapping can be defined. the one-dimensional mapping is formulated explicitly. the generation condition for the chaos is presented in the sense of Li-Yorke. the condition is verified by an experiment corresponding to a circuit example for chaos generation.

Design of three‐dimensional recursive digital filters with symmetry

Abstract: This paper discusses a useful design method for three-dimensional recursive digital filters approximating the desired amplitude response with spherical or cubic symmetry and the constant group delay. the filter structure is chosen so that the denominator is of a separable form, composed of one-dimensional polynomials of the same form, and the numerator is composed of a mirror- image three-dimensional polynomial with 48- hedral symmetry. In the filter design, the denominator and the numerator approximate the phase and the amplitude responses, respectively, which are designed independently. the proposed method is advantageous since the required computation is relatively small and the numerator is determined by solving a linear equation. the stability of the filter is ensured by its structure.

Untraceable electronic funds transfer systems

Abstract: 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.

Sampling bases for the signal space composed of spline functions

Abstract: The signal space composed of spline functions, which are smooth piecewise polynomials and almost free from oscillation, has attracted attention in the problems of interpolation and signal approximation. to make full use of such a signal space, there must be available a clear description of its properties. As a general approach to this problem, it is effective to derive the sampling basis in the signal space. Previously, the authors discussed the signal space composed of spline functions, and derived the sampling basis for the case where the degree is restricted to two. This paper extends the result to the arbitrary degree. Exam ples of sampling bases are shown for the practical cases of the second and the third degrees. the sampling basis derived in this paper provides the basic characterization of the signal space composed of spline functions of arbitrary degree.

Finite element simulation for the design of an ultrasonic cleaning tank

Abstract: This paper describes the application of the finite element method to simulation of an ultrasonic cleaning tank which consists of a liquid bath with vibrating plate walls. the vibration of a plate, from which the ultrasonic radiation takes place, is analyzed without loading, and it is shown that the location where the erosion occurs corresponds to the region of maximum stress in plate bending. Sound pressure distribution in the liquid bath, which is surrounded by the vibrating plate walls, is then considered as a two-dimensional model in which the coupling between the plate vibration and the sound pressure in the liquid is taken into account. Some numerical demonstrations are made, which is compared with the measured results. the technique has again proved to be a useful means of solving problems of this kind.

Optimal design of three‐dimensional FIR digital filters. Design method by linear programming and reduction of computations by exploiting symmetries

Abstract: Recently, the signal processing by the three-dimensional digital filter is considered in such applications as dynamic image processing, medical image processing, robot vision, and analysis of seismic or meteorological data. This paper proposes a method of optimum design for the three-dimensional FIR digital filter using linear programming. the optimum design of the three-dimensional digital filter, including the design by linear programming, requires in general a large computational complexity. From such a viewpoint, this paper discusses the reduction of computational complexity utilizing the symmetry in the frequency response, and the result is actually applied to the design by linear programming. As a result, it is shown, for the case of the spherical lowpass filter, that the computation time is reduced by one order of magnitude by utilizing the symmetry.

Variable threshold soft decision decoding

Abstract: Soft decision decoding is a decoding method which tries to improve the error-correcting capability by utilizing the information concerning the reliability of the received symbol. One of the soft decision decoding methods for practically useful block code, is APP (a posteriori probability) decoding applicable to the majority-logic de-codable code. In APP decoding, the error is decided for each symbol as in majority-logic decoding. This paper proposes a new decoding method (variable threshold APP decoding), in which the error-correcting capability is improved by controlling the order of error decisions in APP decoding. the performance is evaluated by simulation. As a result, it is shown that the variable threshold APP decoding exhibits better characteristics than APP decoding and approximate APP decoding both in bit error rate and block error rate. When, for example, the proposed decoding is applied to (73, 45) difference set cyclic code, a coding gain of 1.0 dB is obtained compared with the approximate APP decoding. In the proposed decoding method, the time required for decoding is increased, while the hardware scale is almost the same as that of the approximate APP decoding.

Robust model matching design method which ensures stability

Abstract: The design method entitled “robust model matching” was recently proposed for the design of practical robust control systems. By this method, if the overall system is stable, any desired robustness in the steady-state and transient characteristics can be guaranteed in spite of parameter variations for minimum-phase plants. However, when using this method it is always necessary to employ a trial-and-error process to ascertain stability. In this paper a robust model matching design method which guarantees stability is proposed for the case when the parameter variations are such that the plant remains a minimum-phase plant without a change in the sign of the gain. Moreover, only the controlled variable and the operating quantity are measurable. By this approach, any desired robustness in the steady-state and transient characteristics can be achieved while guaranteeing robust stability without the necessity of a trial-and-error process. In this paper, first the necessary conditions a robust filter must fulfill to ensure stability are presented. Next, the degree relationships that certain polynomials concerning the model matching controlled must fulfill to design a robust filter are clarified. Further, it is proven that control system stability is ensured by the design of an appropriate robust filter without narrowing the class of realizable reference models. Finally, the proposed design method is set forth and an example design problem is given.

Algorithms for finding internally disjoint paths in a planar graph

Abstract: Three efficient algorithms for finding vertex-disjoint trees and internally disjoint paths in a planar graph are presented. Given a planar graph G and nets consisting of terminals lying on the external boundary, the first algorithm finds vertex-disjoint trees, each of which interconnects all the terminals of a net. the second algorithm determines the maximum number k of internally disjoint paths between two specified vertices in the planar graph G. the third algorithm really finds these k paths. the time complexity of the first algorithm is O(n), those of the second and third are o(n log n). Here n is the number of vertices of the given graph G.

An algorithm to enumerate all rectangular dual graphs

Abstract: Due to the recent development of VLSI technology, circuits have begun to be made on LSI chips in every field of industry. In the layout design, one of the LSI design processes, the main objective, is to minimize the chip area to increase the efficiency and yield rate of ICs. Thus, it is important to find a good floor plan in the sense of area efficiency. This paper considers the problem of enumerating all rectangular dual graphs which are useful in floor plan design of VLSI using the notion of rectangular dual graph. We present an efficient algorithm which runs in O(|V|2) per one rectangular dual graph to be enumerated for a given maximal planar graph G0, where V is a vertex set of G0.

Energy trapping of higher‐harmonic width vibrations in piezoelectric strips and its application to filters

Abstract: The energy trapping of the higher-harmonic width-extensional and width-shear vibrations in a piezoelectric ceramic strip is studied based on the dispersion characteristics of these modes. By the use of energy trapping of the higher-harmonic width vibrations, new small-sized trapped-energy resonators and filters that can be used at the high-frequency range beyond several megahertz are developed. In the resonator and filter using a higher-harmonic width vibration, it is necessary to excite strongly the width vibration of a specific order and to suppress all other vibrations. In this paper, a new electrode configuration is proposed for this purpose by taking into account the stress distribution of the mode. In addition, this method is used for fabrication of 4.5 and 10.7 MHz trapped-energy filters using the 3rd-harmonic width-extensional vibration and the trapped-energy resonators using the 2nd, 3rd- and 4th-harmonic width-extensional vibrations.

Cycle breaking and longest path in channel routing problem

Abstract: Channel routing is one of the automatic routing techniques in LSI design. In the channel routing by trunk-branch scheme, if a cycle exists in the relational graph between subnets, the trunk of a net must be divided to break the cycle. Reference [11] presented a method, which divides the trunk of a net at a terminal on the graph representing the routing requirement. the effectiveness and pseudo-effectiveness of the method are defined. Through the discussion of the necessary and sufficient condition for the realizability of the division, a condition is derived for the routing requirement to be realized. This paper discusses the degree of freedom remaining in the trunk division (i.e., the connection of free arcs in the modification of the requirement graph). the effect of the degree of freedom on the final result, especially on the longest path in the requirement graph, is discussed. As a result, it is shown that if the trunk division satisfies a certain condition, it does not increase much the longest path length.

A priority queue with two priority classes whose nonpriority class call has preemptive and nonpreemptive parts

Abstract: A priority queue with two priority class calls is analyzed. In this model, nonpriority call consists of three parts: preemptive-repeat part, preemptive-resume part, and non-preemptive part. If a priority call arrives during the service of the preemptive-repeat part, the preemptive-resume part and the non-preemptive part, the preemptive-repeat different, the preemptive-resume, and the nonpreemptive priority queueing disciplines are enforced, respectively. As an analytical result, characteristic equations of completion time, busy period, waiting time, system time and through out are derived. Also, from numerical examples, throughput and average waiting time characteristics are examined.

Improvement of convergence characteristics pole-controlled IIR echo canceller

Abstract: 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.

Design of narrow‐band FIR filters using interpolation technique

Abstract: It is an important realization of digital filters to reduce the amount of hardware, notably multipliers. This paper proposes a cascade connection of IFIR (Interpolated FIR) filters by which a steep characteristic can be obtained with a small number of multipliers in most cases. the reduction of the multipliers is achieved by the determination of the transfer function of the overall cascade connection (not each element) by using the Lnorm approximation. This method also has an advantage in that its central frequency can be determined arbitrarily. the cascade connection of several elements reduces the degrading of its characteristic caused by the quantized coefficients as shown in examples, since the conditions required for each element are relaxed in this structure.

Use of generalized cepstral distance measure in isolated word recognition

Abstract: This paper proposes a generalized cepstral discance measure based on minimum phase spictral models and shows the performance of the distance measure for isolated word recognition. the generalized cepstral distance measure approximates the L2 norm between two spectra on a fractional power magnitude scale. the measure is equal to the cepstral distance measure for the special case. Calculation of the distance can be performed efficiently from cepstral coefficients or linear predictor coefficients. Isolated word recognition experiments using a vocabulary of 20 highly confusable Japanese city names indicate that the generalized cepstral distance measure gives higher recognition accuracy than conventional distance measures. Three analysis parameter sets, which are FFT cepstral coefficients, linear predictor coefficients, and improved cepstral coefficients, are also compared in terms of their effects on the recognition performance.

Correction theory and algorithm for reachability matrix model

Abstract: A reachability matrix M is a binary matrix with the reflexive and transitive property, ie, M + I = M and M2 = M, where I is the identity matrix Reachability matrices are used to identify structural models of complex systems While a structural model is being developed or after it has been developed, the developer may want to make changes in it Procedures for changing a structural model with computer assistance are called “correction procedures” Several algorithms for correction procedures of the partially filled reachability matrix are de scribed

A systematic method of synthesizing systolic arrays using a data dependence graph

Abstract: With the recent development of VLSI technology, the systolic array is considered as practically realizable, where a large number of processors are placed on a chip to perform a parallel computation. the systolic array is a dedicated VLSI, which has a hardware structure depending on the algorithm. Consequently, a problem is how to design the systolic array corresponding to the given algorithm. This paper proposes an automatic synthesis for the systolic array using the dependence vector representing the data dependence. It is noted first that the systolic array is determined by the vector along the projection direction and the vector along the computation direction of the data dependence graph. the constraint for the forementioned two vectors is derived for the systolic graph. the constraint for the forementioned two vectors is derived for the systolic array to be realizable. Then using the constraint, the systolic array is derived systematically. A feature of the proposed method is that more than one systolic array corresponding to an algorithm can be constructed without a duplication. the systolic array in which processors operate in different ways, can also be derived. Finally, the proposed method is applied to the matrix multiplication, convolution and LD decomposition, indicating its usefulness.

Real and artificial speech signals for objective quality evaluation of speech coding systems

Abstract: 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.

Analysis of piezoelectric ceramic bending vibrators using interdigital electrodes both for poling treatment and for AC excitation

Abstract: The single-plate ceramic bending vibrator, using interdigital electrodes for poling and excitation, has the remarkable features of high performance and reliability with little spurious response. This is in sharp contrast to the traditional bending vibrators with bimorph structure. This study aims at the description of the design principle for the bending vibrator with interdigital electrodes. A theoretical analysis is made using the conformal mapping, and a detailed discussion is presented on the relation of the capacitance ratio to the electrode dimensions and the poling voltage. Then the optimum condition is derived. A good agreement was observed by comparing the result of analysis and the experiment. It is shown that the capacitance ratio is less for the transversal coupling-type bending vibrator, with the electrode finger being parallel to the direction of elongation, than for the longitudinal coupling type with the perpendicular direction.

Iterative technique for designing IIR filters with arbitrary log-magnitude function

Abstract: This paper proposes an iterative approximation technique of log-magnitude response for IIR filters. Minimization of the mean-square error between a desired log-frequency response and that of an IIR filter leads to a nonlinear problem. In this paper, the least squares problem on the logarithmic scale is approximated by the iteration of a weighted least squares problem on the linear scale. the weighting function is updated using the result of the previous iteration so that the weighted error on the linear scale approximates the squared error on the logarithmic scale. the filter coefficients at each iteration step can efficiently be obtained using a fast recursive algorithm for a set of linear equations. Several design examples are presented, which show that the iterative algorithm converges after one or two iterations in practice and the weighted error at convergence gives a good estimate of the mean-square error on the logarithmic scale. an example is also presented, where both log-magnitude and phase responses are approximated simultaneously.

Application of complex kalman filter to NMR spectroscopy

Abstract: The Fourier transform NMR (nuclear magnetic resonance) spectroscopy is used widely at present as a noninvasive method in the structural analysis of macromolecules in organic chemistry and in the metabolic study of tissue and cells in biochemistry. We believe, however, that the Fourier transform is not adequate in the determination of the density distribution of the nucleus. the NMR signal contains essentially the relaxation (a kind of damping) phenomena, but the Fourier transform does not take this fact into consideration. Consequently, the obtained NMR spectrum is degraded both in intensity and resolution, providing apparent nucleus density distribution which is different from the true value. This paper proposes an estimation of the NMR spectrum, which positively takes the relaxation phenomenon into consideration. the method is derived by extending the Kalman filter to the complex stochastic system and applying the result to the NMR signal represented by the complex stochastic system. the proposed method is applied to the NMR signal of proton (1H) obtained from mayonnaise in 2[T] static magnetic field. the result is compared with the NMR spectrum obtained by the Fourier transform, and it is shown that the resolution and the intensity are improved drastically.

Frequency spectrum estimation method using complex Kalman filter

Abstract: As a method to estimate the frequency spectrum, the discrete Fourier transform is employed widely, since a high-speed algorithm is available. However, the discrete Fourier transform has the following problems. the presence of noise is not considered, and the resolution of the frequency spectrum is determined uniquely by the sample duration. Other problems are that the sampling interval must be kept constant and the decay of the signal is not considered. This paper proposes a high-resolution frequency spectrum estimation where the complex Kalman filter is applied to the observed signal represented by the complex stochastic system, effectively utilizing the a priori knowledge of the signal source. the method is applied to the measured data and to the simulation, and it is demonstrated that a high-resolution frequency spectrum is obtained by suppressing the effect of the discrete approximation of the frequency component, and, when the frequency spectrum is concentrated, the resolution can be improved approximately by a factor of two, without iterating the measurement. It is shown also that the sampling interval can be set arbitrarily in the proposed method, and the noise and the decay can be considered. A theoretical discussion is made on the relation between the special case of the proposed method and the discrete Fourier transform.

Differential companding PCM by using optimum differential bit

Abstract: Two speech coding methods combining dif ferential PCM (DPCM) with near instantaneous companding (NIC) are proposed in this paper. In general, if NIC is applied simply to DPCM, then the waveform cannot be reconstructed in the decoding process because of the bit removal during compression. to solve this problem, a method has been proposed which is based on the accumulation of the removed bits, but the companding ratio in this case is not sufficient for low bit rates. In Method I proposed in this paper the compressed differential data obtained by NIC of the differential data are modified to minimize the error between the original data and the data decoded sequentially, sample by sample, for the given number of compressed bits. Method II is similar to Method I, but an optimum scale factor is chosen from three adjacent values. Thus, the error power is minimized over the block. Segmental SNR is used as a quality measure, and the results of the computer simulation show that higher speech quality is obtained than in the case of first-order predictive ADPCM (ADPCMf). The hardware realization of the proposed methods is extremely simple because both coding and decoding require merely additions and subtractions. This article presents the principle of the methods and the results of the simulation.