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This paper proposes a relaxation algorithm for an inverse filter that is adaptive to environmental change in a multichannel sound reproduction system. For a conventional multichannel sound reproduction system, a design method has been proposed for an inverse filter by means of the Moore–Penrose-type generalized inverse matrix. However, in the inverse filter in the conventional method, background noise and line noise may be enhanced when the room transfer characteristics vary. In order to resolve this problem, an adaptive relaxation method based on the truncated singular value decomposition (TSVD) has been introduced. In the present paper, a microphone for monitoring the noise is placed at a location far from both ears of the listener. Relaxation of the inverse filter is carried out by using the reproduced sound observed by this microphone. From the results of computer simulation, it is found that the proposed method prevents amplification of the noise. Also, as a result of a subjective evaluation experiment on sound reproduction in a real acoustic environment, the sound quality was found to be improved significantly without damaging the sound localization of position. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(7): 15–26, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10159 [ABSTRACT FROM AUTHOR]

In this paper, a formulation of the convergence property of the NLMS method, an updating algorithm for an adaptive Volterra filter, is discussed. To date, much research has been carried out on the convergence property of the updating algorithm in a linear adaptive filter. However, there has been insufficient study of the updating algorithm for the second-order adaptive Volterra filter. In this paper, the convergence property of the E-NLMS method, an updating algorithm in the second-order adaptive Volterra filter, is formulated and the convergence property of the second-order adaptive Volterra filter is theoretically determined. In the formulation of the convergence property we use the first-order recursive filter expression, which is effective for studying various characteristics of the NLMS method in linear adaptive filters. In addition, it is recognized that the nature of the input signal to the diagonal (D) elements of the second-order Volterra system is different from that of the input to the other (ED) components. By means of computer simulation, it is confirmed that the convergence property given by the derived theoretical equation can sufficiently express the actual convergence property. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(10): 40–50, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1124 [ABSTRACT FROM AUTHOR]

gggggAssume that an n × n dot matrix is stored in consecutive n words of a computer with one word being n bit. This paper discusses eight operations to rotate or transpose the dot matrix in horizontal/vertical! slant direction. As is well known, such operations form a group. This paper presents a unified description of the algorithms, and it is shown that all eight algorithms can be derived by suit- ably setting the constants in the unified algorithm. Then it is shown that the algorithms can be transformed to each other by simple transformation rules for the constants, and those transformation rules are shown to form a group. The foregoing algorithms for the dot matrix are not unique, and there can be different descriptions for the algorithm producing the same result. Consequently, there can be various versions of the eight algorithms as well as versions of transformation rules. This paper shows that one of the two groups formed by the transformation rules is isomorphic to the group of algorithms, but the other is not isomorphic. [ABSTRACT FROM AUTHOR]

This paper studies an adaptive filter with a restricted linear sum of the moments of tap coefficients and investigates its applications. In general, when a sampled value filter or a digital filter represented by a surface acoustic filter is realized, the maximum absolute values or the variances of the tap coefficients are restricted. In this paper, restrictions on the maximum values or the variances are imposed by means of a restricted linear sum of several moments around an arbitrary positive number, facilitating the realization of the filter. The number of the moments and the weights are determined in such a way that the frequency characteristics of the filter after restriction satisfy the given specifications. The adaptive system consists of an ideal filter without restriction on the tap coefficients and an adaptive filter placed in parallel. The adaptive filter consists of a transversal filter. Its adaptive algorithm is a gradient algorithm with a small amount of processing, with a restricted linear sum of the moments of the tap coefficients. Since the proof of uniqueness of the converged value of the adaptive algorithm and the upper limit of the loop gain in the algorithm are formulated, the filter designer can easily obtain filter parameters with restricted tap coefficients. Finally, verification of the discussed results is rigorously carried out by computer simulations. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(3): 11–20, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20126 [ABSTRACT FROM AUTHOR]

This paper shows a technique for efficiently analyzing the information flow of a program that generally includes recursive procedures. This technique enables security levels representing the secrecy of data to be assigned according to an arbitrary finite distributive lattice. This paper proves the soundness of the proposed information flow analysis method and shows that this algorithm can be executed in cubic order of the program description size. In addition, a congruence relation is used to extend the algorithm so that functions can be appropriately introduced to enable information within arguments to be hidden. Finally, a prototype system is used to present execution results of the proposed algorithm. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(9): 48–61, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20099 [ABSTRACT FROM AUTHOR]

In this paper, we discuss the performance of M-ary/SSMA and M-ary/SSMA systems using convolutional code and co-channel interference (CCI) cancellation techniques. First, we show a proposed system model and describe a soft decision method of M-ary/SS which can operate soft decision Viterbi decoding. Next, we evaluate the system performance in an AWGN environment and a Rayleigh fading environment by computer simulations. As a result, spectral efficiency of the system using convolutional code is improved in both environments, though its effects depend on the coding rate. Finally, we discuss the systems using CCI cancellation and convolutional code. As a result, when the spreading factor becomes large, its performance becomes inferior to that of the system which applies no convolutional code in an AWGN environment. However, under a Rayleigh fading environment, a considerable improvement of the performance can be obtained. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(6): 11–20, 2002 [ABSTRACT FROM AUTHOR]

In this paper, a new design method is proposed for complex equiripple approximation of an FIR digital filter. In this method, the desired characteristics are varied and the least-squares method is applied iteratively until convergence to an equiripple solution is reached. Although the obtained solution is quasi-optimal, the necessary computational complexity is extremely small in each iteration and the computation error can be kept small. Therefore, the method has the advantage that the design of higher-order filters, which is difficult in the conventional methods, can now be executed at high speed. By computer simulation, the method is compared with the conventional complex Chebyshev approximation method and the effectiveness of the present method is demonstrated. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 84(2): 30–37, 2001 [ABSTRACT FROM AUTHOR]

In this paper, we propose an active noise control (ANC) system without secondary path model (Ĉ filter). The proposed system is based on the simultaneous perturbation optimization method with block process. Consequently, the coefficients of the adaptive filter in the proposed system are updated by error signals only. The conventional ANC system using the Filtered-x algorithm becomes unstable due to the error between the secondary path from a secondary source to an error sensor and its model. On the other hand, the proposed ANC system has the advantage of becoming stable because of not using the model. In this paper, we show the principle of the proposed ANC system and also the effectiveness of this system by computer simulations. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(10): 47–55, 2000 [ABSTRACT FROM AUTHOR]

This paper discusses a method for configuration of a single operational amplifier filter circuit using one operational amplifier (Op Amp) and LCR elements. The proposed single Op Amp filter circuit has a constant resistance input impedance and a zero output impedance. It allows realization of any stable transfer function. In this paper, a method for realization of an allpass function is investigated in particular and its design method is presented. Actually, a computer simulation and an experiment with a test circuit made of discrete components are carried out for a circuit to realize a fifth-order allpass function. The circuit performance is compared with that of a cascaded scheme of Deliyannis circuits. The element sensitivity in regard to the LCR elements, the frequency characteristics of the Op Amp and the sensitivity of the inductor L with respect to Q are derived. It is shown that the proposed allpass circuit has low element sensitivity and the number of necessary reactance elements is identical to the order of the transfer function. [ABSTRACT FROM AUTHOR]

This paper discusses a performance of an adaptive Viterbi decoder, which performs the carrier recovery and the Viterbi decoding simultaneously. The authors have already considered the combined maximum-likelihood sequence estimator (MLSE), according to trellis combined equalization and the Viterbi decoding and showed that the bit-error rate in the fading environment can be improved applying channel estimation respectively for states of the Viterbi algorithm. The adaptive Viterbi decodere discussed in this paper corresponds to the combined MLSE in the case of no intersymbol interference and can easily be extended to the case of the diversity reception. First, The square Euclidean distance is derived approximately by combining the channel estimation by the least mean squares (LMS) and MLSE, and it is shows that the adaptive Viterbi decoder can perform the diversity combining without carrier phase information, with only a little performance degradation compared to the ideal Viterbi decoding with know carrier phase. Then, by a computer simulation, the results of analysis and the numerical are confirmed and the proposed adaptive Viterbi decoder is shown to be effective in the frequency offset and fading environment. [ABSTRACT FROM AUTHOR]

In the arithmetic coding of the multilevel gray-scale image (128 levels) up to now, the fixed model for the gray-scale image is supplied. This paper proposes a method to improve the compression ratio by introducing the minimum description length (MDL) into the encoder to describe more precisely the local property of the image and by adaptively selecting the probabilistic model. In the course of this development, a practical form of MDL criterion is presented to be adaptively applied to the encoding as an approximation to the MDL criterion proposed by Rissanen. A problem in the arithmetic coding of the multilevel gray-scale image using a probabilistic model with a large number of parameters is the deterioration of the compression ratio. To solve this problem, this paper proposes a convergence acceleration method for the average code length for the arithmetic encoding of the high-level gray-scale image. Then, based on the proposed acceleration method, the MDL criterion is introduced into the arithmetic coding of the multilevel gray-scale image. Finally, the effectiveness of the proposed coding method is demonstrated by computer simulation. [ABSTRACT FROM AUTHOR]

Understanding electromagnetic scattering phenomena at high frequencies is important in specifying the location of the local scattered wave (scattering center). This procedure also is available for the practical radar target recognition. Detection of the scattering center also is possible by means of the Fourier transform in the frequency data measured by a network analyzer. However, since its resolution is limited by the measured frequency bandwidth, practical applications may often be difficult. In this paper, the use of the MUSIC algorithm which has a higher resolution than the Fourier transform is proposed for the time-domain estimation method of the electromagnetic scattering. As an example, the detection of the scattering center in the backscattering measurement of a conductive sphere is discussed. It is pointed out that the algorithm can be applied without significant modifications. Most of the super-resolution methods require many snapshots. It is found by computer simulation that the noise suppression preprocessing by time domain gating is effective for reduction of the snapshots. The setting of the number of incident signals (scattering center) of the MUSIC algorithm in the time-gated data is discussed. Further, by experiments, these investigations were verified and the high-resolution nature of the present method is proven. [ABSTRACT FROM AUTHOR]

This paper proposes a genetic algorithm-based method for constructing a fuzzy classification system with fuzzy if-then rules. In the proposed method, a rule selection problem for constructing a compact fuzzy system with high classification power is formulated as a combinatorial optimization problem with two objectives: to maximize the classification rate and to minimize the number of rules. Then a method of implementing genetic algorithms is proposed for the application to this problem and its effectiveness is demonstrated by computer simulations. The implementation of genetic algorithms in this paper employs an approach where a set of fuzzy if-then rules is coded as a single individual. [ABSTRACT FROM AUTHOR]

This paper defines the degree of nonlinearity as a measure for the nonlinearity of the system. A method for estimating the nonlinearity is proposed based on the input and the output time-series signals under the actual condition that the additive observation noise is included. The degree of nonlinearity of the system takes a value between 0 and 1. It approaches 1 when the part of the variational power that carmot be represented by the linear combination of the input increases in the output variation depending on 0-input in the linear system. A multilayer perceptron is introduced as a parametric function which represents the wide class of nonlinear functions needed in the estimation. An example is shown of the family of memoryless and nonlinear systems where the degree of nonlinearity changes from 0 to 1 by the change of the system parameter. An example also is shown of the system with a finite memory having a degree of nonlinearity of 1. By a computer simulation, the validity of the nonlinearity estimation proposed in this paper is demonstrated. The method will be applied effectively to the modeling of the system based on the observed input and output signals. [ABSTRACT FROM AUTHOR]

Vecerbi decoding and sequential decoding are well known as the maximum or approximate maximum likelihood decoding methods for the convolutional code. The decoding error rates of those decoding methods have been examined in the past. However, in those discussions the evaluation has been made by assuming the channel as memoryless. By contrast, this paper proposes several maximum likelihood decoding methods for the convolutional codein the channel model with memory, such as the Gilbert model. In decoding method (I), the conditional probability of the transmitted sequence for the received sequence is determined for each state of the channel. Using the results as the metric, the state of the channel is considered in the same way as the code trellis. The decoding is made by suitably selecting the metric. In decoding method (II), the state of the channel is estimated to suppress the computational complexity with the increase of the channel states. These methods are applied to Viterbi decoding. It is shown by computer simulation that the decoding error rate is improved over the traditional Viterbi decoding with equal complexity in instrumentation compared with the traditional decoding methods. [ABSTRACT FROM AUTHOR]

This paper proposes a method of wave analysis which determines the frequency, initial phase and amplitude of the frequency components from three times the number of data com- pared with the number of components, using only the algebraic method. In the analysis, the frequencies are determined using the formula from sum to product of trigonometric functions, assuming the waveform is composed of a finite number of frequency components. Then using the determined frequencies, the initial phases and the amplitudes are determined. For verification, the signal of the in-band multifrequency signaling system is considered as an example, and a computer simulation was made for the proposed frequency analysis. The signal is composed of one each from four kinds of low and high frequencies. Sixteen kinds of composite waves are constructed. The analysis is made as follows. Variations of ±1.3 percent are given to each of the low and high frequencies, and the initial phase was set as arbitrary in the range from -90 to +270 deg. in regard to the cosine wave. The amplitude is varied from 1 to 1/16. The sampling frequency is 4.48 kHz and the amplitude quantization is made in 12 bit. The maximum error in the result of analysis for the multiple frequency signal is -3.9 percent for 100,000 trials, including the given frequency shift. In the analysis of wave composed of five components, the accuracy of nine digits is realized, using the double precision discrete data. [ABSTRACT FROM AUTHOR]

When synchronization phenomena are analyzed, the phase equations are known to often provide useful qualitative and quantitative information [1]. The target of this paper is to examine through comparisons of the derivation algorithm of the phase equations a realistic system including the case where the equations are not necessarily given explicitly. In other words, the equivalence of two different conventional algorithms [1, 4] is shown, and the advantages and cautions when specific phenomena use both in an example are each explained. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(12): 1– 9, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20284 [ABSTRACT FROM AUTHOR]

This paper proposes a method of realizing a handwritten mathematical formula input system that works in real time. In the handwritten character input, the data is input generally for each stroke (segment). The proposed method is based on this property. The mathematical formula is recognized on the basis of the result of recognition of the individual input strokes and their positional relations. Then the LaTeX source code is generated. In this paper, mathematical symbols are considered as characters, and are integrated into character recognition. More precisely, each time a stroke is input, it is recognized as a single-stroke character. The crossing with the already input characters is examined, and the character is extracted so that the likelihood of forming character is maximized. Based on the positional relations among the extracted characters, the structure of the mathematical formula, such as superscripts/subscripts and fractions, is recognized. The mathematical structures that can be recognized by the proposed method are the structures obtained by combining five different structures, namely, superscript/subscript, root, overline, underline, and fraction. The recognition is not affected by the order of input. The proposed method is applied to various mathematical formulas, and it is seen that the proposed system works adequately as an input environment for real-time processing. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(8): 66–81, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20086 [ABSTRACT FROM AUTHOR]

In this paper, a closed-form expression and a macro model for simulation are proposed for the RLCG transmission line expressed in terms of the telegrapher's equation. First, the hyperbolic functions contained in the admittance matrix of the transmission line are analytically expanded in terms of poles and residues, deriving a close expression of the admittance matrix. Next, from the closed-form expression, rigorous poles of the admittance matrix are obtained. By using the closed-form expression, a macro model of the RLCG transmission line is obtained. The passivity of the macro model, a method for determination of the order, and a simulation method in the time domain are discussed. As a numerical example, a closed-form expression and a traditional discretized model of a transmission line are compared. In this way, the effectiveness of the macro model obtained from the closed-form expression is confirmed. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(4): 1–11, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10155 [ABSTRACT FROM AUTHOR]

In this paper, a new frequency-domain active noise control (ANC) system is proposed in which an optimal step size is used for updating at each frequency. The proposed system achieves convergence faster than in the case of the conventional ANC system with the Filtered-x LMS method updated with the step size converging at the fastest speed. This is because the step size with the fastest convergence can be adapted at each frequency in the proposed system. In the conventional system, convergence is not possible unless the phase error of the secondary path model satisfies the stability condition. In the proposed system, on the other hand, excellent silencing characteristics without divergence can be obtained by letting the step size become 0 at frequencies not satisfying the stability condition, because a different step size at each frequency can be used. In this paper, the theoretical equation for the optimal step size is derived by using only the information obtainable during system operation. Next, a configuration of the ANC system that can make use of the optimal step size derived from this theoretical equation is explained. Further, the method for identifying the frequency causing instability and its control method are explained. Finally, by computer simulation, the effectiveness of the proposed system is verified. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(7): 51–61, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10000 [ABSTRACT FROM AUTHOR]

In this paper, we propose a new method for speeding up synchronous acquisition methods by using code-orthogonalizing filters in DS/CDMA. In the proposed synchronous acquisition method, the normalized complex LMS algorithm is adopted as an adaptive algorithm with code-orthogonalizing filters and can be applied to multipath fading transmission paths. To further shorten the synchronous acquisition time, we study the step size, the number of updates of the tap weights of the adaptive algorithm, and multiple-dwell detection. By simulating these methods on a computer, we evaluated the average synchronous acquisition time as a function of the co-channel power ratio and the number of users, compared the proposed method to conventional methods, and clearly demonstrated its effectiveness. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(6): 38–47, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10051 [ABSTRACT FROM AUTHOR]

Engineering techniques are being vigorously utilized in the medical sector. Examples are the visualization of human body measurements such as CT and MRI by computer graphics, and the prediction of the results of surgery by computer simulation. This paper discusses a system in which a cephalogram is constructed on the basis of anatomical knowledge, and the change in the 3D facial shape after facial surgery, especially corrective surgery on the jaw, is predicted. The proposed model has a layered structure composed of skeleton, muscle, and skin, and can use physical calculations to generate dynamically the modification of the facial surface accompanying the motion of muscle and skeleton. Using this cephalogram as the standard model, the facial surface and the skeleton shape are modified on the basis of 3D coordinate data on the measurement points obtained from two x-ray normalized images taken from the front and the side, and the head model for the individual is constructed. The change in shape of the facial surface when bone is moved following the actual surgical procedure is predicted in 3D. In the prediction experiment, a head model was constructed from the x-ray normalization images of the patient before surgery, and the surgical procedure was simulated. The prediction result was compared with the actual results and evaluated, and demonstrated the effectiveness of the proposed method. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(11): 45–55, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10053 [ABSTRACT FROM AUTHOR]

An extremely significant transition phenomenon from a periodic phase to a chaotic phase is known as torus doubling. In this research, we perform computer simulations to study what kind of periodic behavior exists in the chaotic phase transitioned into due to torus doubling in an electrical circuit that generates torus doubling. The result was the generation of an n-fold torus that is believed to be deeply related to the period of a relatively large window that appears in the chaotic phase in logistic maps such as a 6-fold, 5-fold, or 3-fold torus in the chaotic phase. In this paper, we call the torus generated in the gap of this chaotic phase the torus window. We also analyze a coupled model consisting of a logistic map and a sine-circle map as the model to explain the mechanism for generating the torus window. We perform computer simulations to show that the 6-, 5-, and 3-fold torus windows are generated in this model. © 2002 Scripta Technica, Electron Comm Jpn Pt 3, 85(5): 56–62, 2002 [ABSTRACT FROM AUTHOR]

In this paper, we present a new structure for a complex adaptive infinite impulse response (IIR) notch filter which is useful for the detection of complex sinusoids in communication and radar systems. The proposed structure is based on the gradient algorithm. The adaptive detection of a complex sinusoid with additive complex white noise has been investigated. A quantitative analysis for the steady-state coefficient fluctuation has been developed. Analytical expressions for the steady-state variance of complex variable coefficient fluctuation have been obtained. It has been shown that the variance of the complex coefficient fluctuation does not depend on the angular frequency of the input sinusoid. Finally, the results of computer simulations which confirm the theoretical predictions are presented. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 85(1): 32–40, 2002 [ABSTRACT FROM AUTHOR]

In this paper, a scheme for estimating gas concentration change signals for each gas generating source in a room from output signals of a combustible gas sensor installed in the room, and for estimating event signals synthesized from them by a genetic algorithm (GA), is proposed. It is shown by computer simulation that more accurately estimated concentration change signals or event signals are obtained by applying selective weeding, crossing, mutation, and so on, repeatedly to an individual group, while applying an appropriate evaluation function to concentration change signals and event signals estimated for each gas generating source expressed by a GA individual. In addition, it is shown that the estimated concentration change signals and estimated event signals obtained from the output signals of a gas sensor installed in an occupied room by the proposed scheme correspond approximately to human activities. In addition, it is shown that the proposed scheme is effective in detecting human activities with a gas sensor, since appropriate signals can be estimated robustly even though components such as noise may be present in the sensor output signals. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 85(1): 50–61, 2002 [ABSTRACT FROM AUTHOR]

When high-speed transmission is carried out for multimedia mobile communication, it is indispensable to overcome the effect of intersymbol interference due to time-varying multipath propagation. As one of the remedies, the time domain optimum receiver using an adaptive array antenna has been proposed [1]. The time domain optimum receiver consists of three adaptive signal processing sections: an adaptive array antenna, a whitened matched filter, and a maximum likelihood estimator. However, discussions on their adaptive updating methods have not been sufficient. In this paper, centered around adaptive updating, the configuration of the adaptive spatial and temporal optimum receiver and the updating method are discussed. First, by means of the array antenna, the direction of arrival of the received signal is estimated. Separation of the directions of arrival is carried out by adaptive beam forming. By means of the matched filter, the delay profile for each direction of arrival is obtained. Such a configuration and an adaptive updating method are described. A method for improving the metric calculation of the propagation path estimator and trellis in the maximum likelihood sequence estimator (MLSE) by Viterbi algorithm is also presented. By computer simulation, improvement of the convergence characteristics of each adaptive controller and bit error rate is shown. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(5): 23–32, 2001 [ABSTRACT FROM AUTHOR]

Blind signal separation is a signal processing technique that can extract the original signals from their mixtures by using only the observed signals when the mixed signals can be observed by multiple sensors. In this paper, we propose a technique for blind signal separation of separating original signals from the observed signals which mix the nonstationary signals, such as speech or music. The proposed method implements blind signal separation by minimizing a nonnegative function that achieves the minimum (zero) when the second-order cross-correlation value of the observed signals becomes zero. The effectiveness of the proposed method is verified by computer simulations and tests that used a mixed speech signal observed in an ordinary room. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 84(2): 1–11, 2001 [ABSTRACT FROM AUTHOR]

This paper proposes a new path diversity method using an adaptive array antenna in order to improve the quality and capacity for high-data-rate land mobile communication systems. In the proposed scheme, some elements from circularly arranged directional antenna arrays are selected based on estimated delay profiles, and some outputs of the adaptive array are combined for each delayed wave. Therefore, interference waves can be suppressed by this space-time processing approach with low computational complexity. Computer simulations show that the performance of the proposed scheme can be improved drastically by the path diversity combining method which minimizes mean square errors, even if delayed waves are incident from the same directions as a direct wave. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(12): 9–18, 2000 [ABSTRACT FROM AUTHOR]

This paper proposes an adaptive second-order IIR bandpass decimation filter for detecting a single sinusoid with an unknown frequency embedded in noise. The purpose is to realize a filter with high Q (selectivity) and less output amplitude fluctuation after convergence. As the first step, the adaptive second-order IIR bandpass filter with the variable center frequency is outlined. Then, we present a new method that derives the decimation filter function with a small amount of computation. The filter is derived from the above filter with the denominator coefficient made variable, and with the polynomial of z-M as the denominator, where M (integer) is the decimation ratio. A method is shown to design the adaptive signal generator and the coefficient update circuit for a single sinusoid detection adaptive filter based on the proposed method. Lastly, the convergence and the output error characteristics of the proposed adaptive decimation filter are examined by a computer simulation example. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(7): 91–101, 2000 [ABSTRACT FROM AUTHOR]

This paper discusses the moving obstacle avoidance problem using intermittency chaos. It is a proposal of modeling an agent-oriented robot, which controls the orbit by utilizing chaotic fluctuation. More precisely, a modified Bernoulli system that generates intermittency chaotic mapping is considered. By varying the parameters of the system, a wide range of controls, from random-rich control to locally converging control, can easily be realized, which is expected to result in more adequate orbit control. This paper discusses the modeling of the orbit control by the above method for the moving obstacle avoidance of the robot. The effectiveness of the method is evaluated based on the results of simulation. The following observations are made. (1) Efficient orbit correction can be realized, where the obstacle is avoided by the path that minimizes the loss in regard to the moving distance and the energy consumption. (2) Even in a complex environment composed of multiple obstacles, adequate avoidance can be realized by a simple algorithm. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(6): 19–29, 2000 [ABSTRACT FROM AUTHOR]

ln the design of integrator-based filters, there are degrees of freedom in selection of the signal flow graph representing the connection state of the integrator. In this paper, a procedure is proposed for selecting the signal flow graph for a filter with a small deviation from the ideal frequency characteristics. In general, the integrator works ideally near the unit-gain frequency. Hence, in order to realize a filter with excellent characteristics, a guideline is described for selecting the signal flow graph such that the unit-gain frequencies of all integrators used in the filter are close to each other. Since the amount of computation for selection of the signal flow graph becomes enormous even in a filter of low order, an evaluation scale for search is used that can easily estimate the spread of the unit-gain frequencies in individual signal flow graphs. As examples, the signal flow graphs are derived for the fifth-order maximally flat-delay low-pass filter and the third-order amplitude equal-ripple low-pass filter. The validity of the proposed evaluation scale is presented. The effectiveness of the design guideline is confirmed by a computer simulation of the filter made up of MOSFET based on the obtained signal flow graphs. © 2000 Scripta Technica, Electron Comm Jpn Pt 3, 83(6): 61–68, 2000 [ABSTRACT FROM AUTHOR]

Soft decision decoding is a decoding method which can cut the decoding error probability as compared with hard decision decoding by utilizing the channel measurement information effectively. In this paper, a new soft decision decoding method by which decoding operations can be efficiently carried out by utilizing the hierarchy of linear block codes is proposed. In addition, an improved algorithm which can reduce significantly the maximum computational complexity is also proposed. The proposed decoding algorithm utilizing the hierarchical structure of codes is a decoding method by which decoding is carried out sequentially from codes with high coding rates or codes of higher hierarchy in the inclusion order, and soft decision decoding can be efficiently carried out by utilizing a number of classes of codes having inclusion relationships in a manner which is superior to Chase algorithm 2 in the decoding error probability and the computational complexity for decoding. Computer simulation results confirm that the proposed algorithm realizes decoding error probability characteristics close to those of the maximum likelihood decoding method and that decoding can be carried out efficiently at a lower computational complexity than Chase algorithm 2 when the SNR is large. In addition, the improved algorithm which can reduce significantly the maximum computational complexity can realize decoding error probability characteristics close to those of the maximum likelihood decoding method on practical communication channels with relatively large SNR. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(3): 108–114, 2000 [ABSTRACT FROM AUTHOR]

Mean-field Boltzmann machine learning is recognized as a practical method to circumvent the difficulty that Boltzmann machine learning is very time-consuming. However, its theoretical meaning is still not clear. In this paper, based on information geometry, we give an information-theoretic interpretation of mean-field Boltzmann machine learning and a clear geometrical explanation of the approximation used there. Based on this interpretation, computer simulations for evaluating the effectiveness of mean-field Boltzmann machine learning are carried out for two-unit Boltzmann machines. The necessity of geometrical analysis in demonstrating the effectiveness of mean-field Boltzmann machine learning is discussed. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(8): 30–39, 1999 [ABSTRACT FROM AUTHOR]

The geometric learning algorithm (GLA) was proposed as an application of the affine projection algorithm for an adaptive filter to the perceptron. In the GLA, the connection weight vector w(n) is updated vertically toward the orthogonal complement of k patterns. The GLA demonstrates some typical behavior when the learning rate λ is 2, which means that w(n) and w(n + 1) are symmetric with respect to the complement. Therefore, in this paper, the GLA with λ = 2 is discriminated as the symmetric learning algorithm (SLA) and the convergence properties of the SLA are analyzed. The convergence conditions regarding the order k of the SLA, the number P of patterns, and the dimension N of patterns are analyzed theoretically. It is proved that k < N is the necessary condition for convergence when P > 2N. The relation between k and the learning speed is analyzed theoretically. It becomes clear that the maximum learning speed on average can be obtained when k = N/2. These properties are supported by computer simulations. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 82(4): 18–25, 1999 [ABSTRACT FROM AUTHOR]

In this paper, an adaptive array is proposed in which the CMA (Constant Modulus Algorithm) and LMS algorithms are combined so that a pass diversity effect is obtained without using training signals. The basic operations of the proposed algorithm are as follows. CMA captures the direct wave with maximum power. Based on the output of the CMA adaptive filter, a reference signal is generated. The LMS algorithm, using this reference signal, captures the delayed wave. By combining the direct wave and the delayed wave at maximum rate, the receiving quality is improved. By means of a computer simulation, we confirmed that the convergence property is improved and that the effect of the reference signal error on the system convergence characteristics is small. © 1998 Scripta Technica. Electron Comm Jpn Pt 3, 81(7): 37–44, 1998 [ABSTRACT FROM AUTHOR]

A novel adaptive filter is proposed that adapts to the sample spacing as well as to the tap weights. Although in adaptive digital processing, the tapped-delay line usually has uniformly spaced taps, with the spacing equal to symbol period or a fraction of symbol period, an adaptive filter with a few taps should be adaptively arranged for nonuniform spacing for better convergence as the sampling timing varies. The conventional non-uniformly spaced filters are not adaptively changed or not arranged under the constraint of the acceptable mean-square error. In this paper, a least-mean square (LMS)- type algorithm is derived in which an adaptive filter adaptively changes the amount of delay, as well as the filter coefficients. The performance of the proposed algorithm is investigated by computer simulation. [ABSTRACT FROM AUTHOR]

Learning identification is one of the most widely utilized adaptive algorithms due to its small computational complexity. A problem with this algorithm is that a division by the square norm of the state vector is includ- ed in the coefficient update process which easily produces an instability when the nonstationary such as speech is given as the input. A method known to cope with such a problem and to realize the stable convergence involves not updating the coefficient when small input signals continue. This paper discusses this approach. As a first step, the situation is assumed where the square norm of the state vector is less than the specified threeshold and the effect of interrupting the coefficient update is analyzed. The convergence and the time-constant are formulated and the guarantee of the estimation accuracy for the convergence value is discussed. Then, the step gain is derived for the specified guarantee value so that the convergence speed is the fastest in the stochastic sense while ensuring stability. Lastly, the effectiveness of the proposed method is verified by computer simulation. It is shown that, even if the input signal is annotation, the fastest convergence speed is realized, together with the satisfactory estimation accuracy, which is always obtained according to the specified guarantee value after the convergence. [ABSTRACT FROM AUTHOR]

This paper proposes the parallel construction of the second-order infinite impulse response (IIR) adaptive digital filter (ADF) as an adaptive digital filter for eliminating multiple narrowband signals mixed in the broadband signal. Then the pole-control method for the filter is proposed. In the proposed pole-control, the pole radius and the pole argument are controlled separately by different algorithms. In this study, the pole argument is corrected by estimating the deviation of the central frequency of the signal to be eliminated from the convergent value of the multiplier coefficient that determines the zero. The pole radius is controlled by the method that determines the direction of correction, using the parallel-connected three ADFs with the same pole argument and a slightly different pole radius. It is verified by a simulation that those pole-control methods work effectively. It is demonstrated that the pole radius and the pole argument can be controlled simultaneously. [ABSTRACT FROM AUTHOR]

Orthogonal fractal basis functional sets are proposed for the compression of images with long-tail power distributions. These functional sets are generated by the method of midpoint displacements from deterministic theory and are orthogonalized by the method of Schmidt. From computer simulations of transform data compression coding that uses a transform based on these functional sets and a nonuniform quantizer, the functional sets and a nonuniform quantizer, the functional sets proposed in this paper are shown to better preserve the self-similarity of fractal image data when compared to conventional methods that use the discrete cosine functional (DCF). Also, for standard images that possess local and approximate self-similarity fractal characteristics, although the measured SNR performance is lower, the proposed functional sets are preserve the fractal properties of the images. On the other hand, for standard images that do not possess these fractal properties, compression errors are apparent in blocks from both smooth and edge regions of the images. [ABSTRACT FROM AUTHOR]

In this paper, a new algorithm for the detection of a single sinusoid corrupted by Gaussian noise using IIR adaptive bandpass/notch filter is presented. As the first step, the relation between the convergence speed and the filter Q-factor in IIR adaptive bandpass filter is analyzed. Based on the result of analysis, an algorithm is derived in which the filter Q-factor and the adaptive control signal generator Q-factor are made continuously variable so that they are inversely proportional to the distance between the filter center frequcncy f0 and the frequency f of the sinusoid to be detected. This makes both the convergence speed of f0 and f and the resolution high. Then it Is found that limiting the maximum amplitude response of the adaptive control signal generator to a value proportional to sin 2&pie;f can suppress the coefficient fluctuation after convergence effectively. By this approach, an IIR adaptive filter with a high Q can easily be realized. As the next step, an additional high-speed and highly stable adaptive algorithm is proposed in which the step size of the filter coefficient update is made to vary inversely with the fitter Q-factor. A realization of the IIR adaptive notch filter is also presented. Finally, the usefulness of the proposed adaptive algorithm is verified by a computer simulation. [ABSTRACT FROM AUTHOR]

This paper presents a least-squares adaptive equalizer that provides good performance even in al ill-conditioned environment where the channel distortion is rather severe. Using complex-valued tap coefficients and halving the order of the transversal equalizer result in a substantial reduction in the condition number of the input correlation matrix composed of input signals to the equalizer. The input and output signals for the equalizer are transformed into analytic signals. Then a sampling operation of decimation by a factor of two is performed. Adaptive filtering is performed based on complex arithmetic. Compared with commonly used real-valued adaptive equalizers, accuracy of the estimation of tap coefficients is improved substantially and the computational burden per time is reduced. Computer simulations will show the effectiveness of the proposed adaptive equalizer. [ABSTRACT FROM AUTHOR]

When examining a random group of scatterers, random granular patterns called speckle appear on echograms obtained by pulse echo ultrasonic diagnostic equipment. In living body organs such as the liver, there is a profound connection between the irregularity of the tissue structure and the severity of the disease. Therefore, if the degree of the irregularity can be estimated through analysis of the echogram, the possibility exists that the properties of the tissues can be diagnosed quantitatively. This paper considers a medium that grouped scatterers as the model of the object end investigates the relationship between the statistics of interference pattern such as the speckle that appears on the ultrasonic echogram and the irregularity of the scattering medium. Also examined is the possibility of estimating the degree of irregularity of the medium from the echogram. From the results of a computer simulation, it is found that if the density of the scatterers is known, the degree of irregularity of the medium can be determined from the value of the signal-to-noise ratio (SNR) statistic of the echogram. In addition, it is learned that by using the SNR as the statistics together with the skewness of the brightness histogram of the image, it is possible to estimate the degree of irregularity and the density of the scatterers. [ABSTRACT FROM AUTHOR]

The advantage of the multilevel coding scheme is that the coding can be performed independently for each level, and the optimal coding can easily be determined for the channel with superposed noise with arbitrary probability density distribution. To evaluate the performance of this coding scheme, an approximate formula for the bit error rate after decoding is derived. As an example, the channel with the impulsive noise is considered. Using the approximate formula, the error-correcting capability is assigned optimally to the levels and the coding gain of 20 dB is obtained for the BCH code of code length 31. The foregoing result is confirmed by a computer simulation. A report has already been proposed for the case where the errors of the decoder outputs are randomized except for the top level, by interleaving among the levels. However, a problem arises in that the delay as well as the circuit scale is increased. From such a viewpoint, this paper discusses the case without interleaving. Finally, the performances are compared between the cases with and without interleaving among levels. [ABSTRACT FROM AUTHOR]

The digital sinusoidal oscillator considered in this paper is one of the signal sources employed in the digital signal processing system. A construction of the oscillator is discussed where the memory capacity is reduced and the increasing rate of the noise is suppressed by interpolating the output of the oscillator of table look-up type or feedback type. A design method for the interpolation filter is presented first where no interpolation error is produced if the coefficient wordlength of the interpolation filter is infinite. Then, considering the case where the wordlength is kept finite, the hold-type sampling expander is proposed where the waveform distortion is kept low compared with the traditional zero-interpolation-type sampling rate expander. The distortion in the proposed method is analyzed where the wordlength of the interpolation filter coefficient is kept finite, and the waveform distortion is shown to be low. The increasing rate of the round-off error is examined by a computer simulation for the oscillator, which is obtained by applying the interpolation to the feedback-type oscillator. The effectiveness of the interpolation is demonstrated. Finally, a construction of the block-digital sinusoidal oscillator is shown with a smaller hardware complexity than in the traditional method, by constructing the block-structured oscillator by realizing the proposed interpolation filter as a polyphase filter. [ABSTRACT FROM AUTHOR]

With the recent development of large-scale, high-level functions and high-speed VLSI circuits, intensive research and development efforts have been made to obtain new circuit simulation techniques. This paper proposes a cooperative development environment, which is indispensable in developing the next-generation circuit simulation technique. Assuming UNIX and C language, a proposal for the following scheme is presented. Various circuit simulation functions are realized as independent programs (simulation commands) as far as possible. By giving specifications for those commands in a unified way, data are shared among different simulation techniques. The input file for the simulation command is constructed by filing three kinds of data: circuit connection data; the device model data; and the data for the control. The core file is used as the output file. Then the function and realization of the simulation shell, which is to execute the simulation command interactively or to efficiently share the data among the commands, are discussed. Finally, the method of sharing the circuit simulation program using the shared library function is proposed. Experimental examples are presented for the simulation commands such as shared library function and transient analysis. [ABSTRACT FROM AUTHOR]

This paper proposes an efficient method to detect the hazard existing in the logic circuit using the five-valued logic simulator. The proposed method is effective in logic verification for the combination and asynchronous and sequential logic circuits as well as the verification for the hazard (race). Using the proposed method, the timing of the logic circuit is analyzed, including the rise/fall delay considering the signal transition time. The hazard is detected based on the output of the gate and the scheduled time. Also discussion is a method of event cancellation, which is a problem in the delayed simulation. Through the results of experiments using several circuits, the proposed algorithm is shown to be useful. [ABSTRACT FROM AUTHOR]

An adaptive digital. filter (orthogonal function-type ADF) composed of an orthogonal function system which can locate poles of a transfer function in arbitrary positions in the unit circle in the z plane is known as a way to improve convergence speed in the case in which input is a colored signal (such as voice). However, no sufficient theoretical investigation has been done on the reason why this ADF rapidly converges to the colored signal input. This paper derives an equivalent circuit of an orthogonal function-type ADF in the case in which poles of a transfer function are located in equal space on the unit circle in the z plane. Then the convergence speed using this equivalent circuit is considered. It is shown that the orthogonal function-type ADF is equivalent to a construction due to a periodic process of an input of the known ADF using the frequency sampling filter (frequency sampling-type ADF). In other words, even if Input is colored like voice and aperiodic process, in the orthogonal function type it is shown that orthogonality between tap coefficient inputs is preserved. Furthermore, these studied results are verified by computer simulations. [ABSTRACT FROM AUTHOR]

An orthogonal function-type adaptive digital filter (orthogonal function type ADF) is known as a means of improving the convergence speed of a tap coefficient in the case in which input is a colored signal such as a voice. Two different methods have been used for pole location of a transfer function of this ADF. One is the pole location of equal interval of 2π/N(rad) (N: tap number of ADF) starting at the phase 0 (rad) on the circle whose radius is very close to 1 within the unit circle of the z-plane. The other is the one resulting from rotating the above location by π/N(rad). In the case of the former pole location, it has been shown that it is equivalent to a frequency sampling type ADF such that input is a periodic process of period N. Why improvement in convergence speed of the latter is the case is not clear at this date. in this paper we first derive an equivalent circuit of an orthogonal function type ADF using the latter pole location. In this circuit a modified discrete Fourier transform (MDFT) appears in which phase of a pole of a resonator of frequency sampling filter is shifted by K/N(Iad). Next, we present a condition of expansion coefficients of the MDFT circuit to be orthogonal to one another. Moreover, we clarify that this equivalent circuit is of a construction that satisfies the orthogonal condition of the NDFT circuit, Finally, we verify these results of investigation by several computer simulations. [ABSTRACT FROM AUTHOR]

One of the most important measures for the construction of the active RC filter is the element sensitivity. One of the effective means to reduce the element sensitivity is the method to simulate the conventional LC filter with both ends terminated by resistors, This paper considers the realization of the low-pass filter by I,C simulation using GIC or by the node-voltage simulation. The prototype LC filter is discussed to solve the bias instability problem, which is a problem in the realization. First, an LC filter is proposed which is obtained by adding an inductor in parallel with the input resistor of the conventional doubly terminated LC filter. The method of constructing the filter is discussed. Then a method is presented to simulate the filter using GIC. Finally, the 5th-order Chebyshev low-pass filter is constructed and its characteristics are measured. The element sensitivity is analyzed using a computer, and it is verified that the proposed circuit has a low element sensitivity. [ABSTRACT FROM AUTHOR]

The output noise in the cascade switched-capacitor (SC) filter depends greatly on the pole-zero combinations as well as the order of cascading the second-order circuits. This paper derives the general condition for the combination/order so that the output noise is minimized. First, the noise transmission from each noise source to the output is analyzed for the second-order circuit. It Is shown that the sample-and-hold component in the output noise Is sufficiently large compared to the continuous-time component, and the effect of the pole-zero combination is small. Then, the condition for minimizing the noise transmitted from each second-order circuit to the output of the SC filter is investigated. The derived condition is that the amplitude response of the transfer function from the filter input to each second- order circuit should be flat. Translating the condition into the condition for the pole-zero combination/cascading order, the following condition is derived. For the combination, starting from the pole with higher Q, the closest zero is combined; for the order, the circuits are cascaded so that the frequency at which the amplitude response from the filter input to the second-order circuit should be distributed. Finally, simulations are executed for 1 lth-order low-pass and high-pass filters. It is seen that for approximately 40 percent of 36 ways of combination/order with the foregoing condition, the output noise ranges within 20 percent of the minimum value, which indicates the effectiveness of the presented condition. [ABSTRACT FROM AUTHOR]