bluuebunny/crossref_metadata_2025_split huggingface dataset:
Asynchronous excitation phenomena in oscillatory circuit containing periodically operating analog switch
Authors: Matsuki Michitaka , Mori Shinsaku 
Published: July 2007

AbstractIn an oscillatory circuit containing a nonlinear negative resistor whose v‐i relationship is i =‐g1v + g3v3, asynchronous excitation by external force cannot exist under certain conditions. However, it can be confirmed experimentally and theoretically that excitation can exist when the resistor is biased and periodically switched by an analog switch at a much higher frequency than both resonant and external frequencies. The oscillation modes obtained change in the order of trivial, hard, soft, hard and trivial as external force increases. This mode variation has been found to be the same as that in i = g, 1v ‐ g3v3 + g5v5 except for the hard oscillation.
Chaos and fundamental bifurcation phenomena from a relaxation oscillator with periodic thresholds
Authors: Kohari Ken'Ichi , Saito Toshimichi , Kawakami Hiroshi 
Published: July 2007

AbstractA relaxation oscillator which includes a hysteresis element with periodic variation of the threshold is analyzed. Deriving a one‐dimensional return map, the parameter regions are clarified where the system exhibits a periodic solution or quasi‐periodic solutions and where the response characteristics of the system having periodic solutions becomes devil's stair and where the attractors may coexist. When the system has chaotic response, then coexistence of chaotic and periodic attractors is confirmed. Various results are confirmed from the laboratory experiments.
A synthesis of an optimal fuzzy filter based on local statistics
Authors: Takashima Hironori , Taguchi Akin , Murata Yutaka 
Published: July 2007

AbstractThis paper presents a design method of data‐dependent filters that uses simplified fuzzy inference. Since the antecedents of fuzzy inference can comprise several local characteristics (i.e., observation values), it is possible for the fuzzy filter to adjust its weights to adapt to local image data. the tuning of membership functions and fuzzy rules of the proposed filter results in a least mean square (LMS)‐like algorithm. Thus, local characteristics can be increased for the proposed fuzzy fiter optimally. This paper, introduces a new observation value (calculated from local statistics) into the proposed filter. the proposed filter changes filter behavior according to the local properties of signals and provides good noise attenuation in all regions of image, including detail regions, while still preserving the details.
Nonlinear signal processing based upon a reconfigurable knowledge database
Authors: Iiguni Youji 
Published: July 2007

AbstractOne of the most important problems in signal processing is to estimate the output for a known input, here referred to as a query from the input/output (I/O) data seen to this point. A nonlinear adaptive estimation method that uses a k‐d trie (k‐dimensional binary digital search tree) as an alternative to the k‐d tree is presented here. The observed data are stored in the k‐d trie, and a local model to answer each query is built by using thennearest neighbors of the query. The database contents are updated as each new data point is available such that the latestNdata are contained. This enables nonlinear adaptive signal processing in nonstationary environments.A storage procedure for the k‐d trie that allows efficient update is presented for further reduction in computation time and storage requirement. A recursive least square algorithm, used for a recursive production of the local model, improves computational efficiency. Finally, a method of removing outliers is presented for improving estimation accuracy.
A partial realization algorithm for finite‐size two‐dimensional arrays
Authors: Matsuo Tsuyoshi , Hasegawa Yasumichi , Okada Yoshikuni 
Published: July 2007

AbstractThe following figure realization problem has already been posed. “For arbitiarily given figure (two‐dimensional array), find the (mathematical) model that realizes (describes) the figure. If there exist two or more models, show that they are isomorpric.” As a solution to the problem, the figure realization theorem is obtained as follows. “For any given figure, there always exists a canonical commutative linear representation system that realizes the figure, which is unique except for the isomorphism.” There also are studies to analyze the finite‐dimensional canonical commutative linear representation system.Based on those results, this paper discusses the partial realization (faithful description) problem for the finite‐size two‐dimensional array. an algorithm is given which derives the minimum‐dimensional commutative linear representation system that partially realizes (faithfully describes) the given finite‐size two‐dimensional array. As an application example, the dyeing and weaving patterns are considered and the usefulness of the proposed algorithm is shown. the proposed method is a processing to compress the information of the finite‐size two‐dimensional array by maximally utilizing the regularity of the finite‐size two‐dimensional array considering the vertical and horizontal positions.
Optimum linear transformation for a quadratic form of signals
Authors: Sakamoto Hiroyasu 
Published: July 2007

AbstractThis paper proposes the optimum skew linear transformation of signals which gives the minimum mean error of a positive definite quadratic form of the signals. the relation of this problem with hybrid coding is emphasized. Based on conventional orthogonal KLT, the condition for the optimum transformation is given in a generalized eigenvalue problem. We show the geometrical meaning and a solving method of the condition, and discuss the existence and the uniqueness of its solution. Numerical performance comparisons show superiority of this transformation to others such as KLT and DCT. Because this transformation is a generalization of KLT to a general quadratic form, it is as significant as KLT is.
Prefiltering for blind equalization
Authors: Itoh Katsuko , Shimamura Tetsuya , Suzuki Jouji 
Published: July 2007

AbstractA major problem with blind equalization algorithms based on the distribution matching principle is that they need a long time to accomplish their convergence. What affects convergence has not been discovered to date. the similarity between the least mean‐square (LMS) algorithm (widely used for adaptive equalization) and the blind algorithms is considered here. It is expected that the convergence of the blind algorithms is dependent on the condition number of the correlation matrix of the input sequence. Prefiltering methods, including coefficient‐fixed type and coefficient‐adaptive type, are derived for blind equalization. In these methods, the prefilters are realized by a prediction error filter, which has the ability to compensate for amplitude distortion induced by a channel. Since the prefilters have such ability, the blind equalizers to be cascaded with them are required only to compensate for phase distortion. As a result, the burden imposed on the blind equalizers is reduced. the prefilters output a near‐white sequence and lead to an improvement in the convergence of the blind algorithms, being degraded as the condition number is increased. the effective ness of the proposed methods is validated by computer simulations.
Design of noise generator using chaotic circuit
Authors: Suzuki Katsunori , Mori Shinsaku , Nishio Yoshifumi 
Published: July 2007

AbstractThis paper proposes an approach for a noise generator based on the chaotic circuit. In the proposed circuit, the diode is assumed to be an ideal switch. By means of this simplification, it is possible heuristically to understand the generation of several oscillation frequencies in the circuit. Also, it is possible to confirm the switching of the frequency from the attractor. The frequency depends strongly on the memory device in the circuit. More frequencies are generated by increasing the dimensions of the circuit since the spectral characteristics can be controlled easily by the variation of the circuit elements and by the increase of the dimensions of the circuit. Hence, the structure suggests an application to a noise generator. Further, the validity of the analysis results are confirmed by the circuit experiment.
Design of a cerebellar model arithmetic computer with adaptive resolution and its application to nonlinear signal processing
Authors: Kawamoto Isao , Iiguni Youji , Adachi Norihiko 
Published: July 2007

AbstractA cerebellar model arithmetic computer (CMAC) with adaptive input space quantization is designed. the quantization intervals are compressed by the segmentation of weights in the region where the function to be learned changes rapidly, whereas those are extended by the integration in the region where the function changes slowly. the adaptive resolution technique reduces required memory space and improves learning speed. Results of simulation studies on nonlinear signal processing problems are reported. A storage procedure for the weights and new criteria for the integration and segmentation are also reported.
Mathematical programming formulation for neural combinatorial optimization algorithms
Authors: Urahama Kiichi 
Published: July 2007

AbstractThis paper considers the anlog neural solution of the combinatorial optimization problem. the solution method is analyzed based on the Lagrange multiplier method for the continuous relaxation problem of 0‐1 integer programming. It is shown that the solution process can be interpreted as the solution by the gradient method for the saddle point of the Lagrange function. an improved Hopfiled net is derived from the formulation as the pure integer programming. the elastic net and the generalized deformable model are derived from the mixed integer programming problem. Based on those results, an interpretation of the deterministic annealing is derived from the viewpoint of mathematical programming. It is shown that the Lagrange function can work as the Lyapunov function for the solution process and the convergence property of those neural methods of solution is analyzed.
Vias‐per‐net in multilayer multiterminal topological routing
Authors: Fujiyoshi Kunihiro , Kajitani Yoji 
Published: July 2007

AbstractThe proble of finding routing and layer assignment, which minimizes the number of vias ignoring the line width and the separation of lines, is NP‐hard even if the nets are restricted to two‐terminal nets. Numerous heuristic algorithms have been developed in an attempt to improve efficiency. However, as for properties of solutions, it has only been proven that the number of vias for t‐terminal nets in an optimum solution (min vias) is at most t ‐ 1, if the number of layers is restricted to 2. It is also known that properties of vias and terminals can be defined in a number of different ways for the multilayer topological via minimization (TVM) problem using unlimited number of layers. the problem becomes complex. In this paper, the forementaioned theroem is generalized and it is proven that a t‐terminal net uses at most t ‐ 1 adjacent‐layer connecting vias in any optimum solution for the general case without any restriction on the number of layers. Further, a class of problems is shown to exist which require that many vias. Similar results are also shown for all‐layer connecting vias.
A method of measuring moving characteristics of precision mechanical‐translation stages using ultrasonic plane waves and applying it to a line‐focus‐beam acoustic microscopy system
Authors: Ono Yuu , Kushibiki Jun‐Ichi , Chubachi Noriyoshi 
Published: July 2007

AbstractA method of measuring moving characteristics of precision mechanical‐translation stages using ultrasonic plane waves is presented. The moving characteristics are obtained by analyzing the phase variation of ultrasonic waves reflected from a reflector set on the translation stage. The resolution is 4 nm at an ultrasonic frequency of 225 MHz; higher resolution is expected at higher frequencies. The method is demonstrated for a line‐focus‐beam (LFB) acoustic microscopy system. Measurement errors of the phase velocity of leaky surface acoustic waves, caused by the positioning errors of thez‐stage, are investigated both theoretically and experimentally. The linear positioning and straightness errors of thez‐stage are estimated at 225 MHz and these effects on velocity measurements are successfully removed using the measured moving characteristics.
Speaker identification using multipulse excited linear prediction analysis (LPC) vocal tract model
Authors: Hangai Seiichiro , Miyauchi Kazuhiro , Urabe Nobuyuki 
Published: July 2007

AbstractThe authors devised a method to identify a speaker where feature parameters are derived by normalizing the position and the magnitude of the subpulses in the multipulse train extracted from the vowel and are combined with 14th‐order linear prediction analysis (LPC) cepstrum. In other words, 10 code vectors are constructed for each registered speaker in the 16‐dimensional space. the speaker is identified by determining the code with the smallest VQ distortion for the vector of the unknown speaker.A speaker identification experiment is executed using five sustained vowels of 100 speakers for the same time period. A 100‐percent identification rate is realized by combining the feature parameters of the multipulse source composing the feature vector and the feature parameters of the LPC cepstrum by the weights of 3:2. the speaker identification also is attempted for 10 subjects after 10 days by the same method, and the identification rate of 69 percent is obtained. the result is an improvement of 12 percent, compared to the case where only the LPC cepstrum is used. the identification rate also is improved for the speech one and two months later.
A new type circuit of switched‐capacitor transversal filter using transfer function coefficient ratios
Authors: Kawakami Izumi , Mori Toshihiro , Takahashi Nobuaki , Takebe Tsuyoshi 
Published: July 2007

AbstractThe switched‐capacitor (SC) filter is considered to be a very practical one for integrated circuit (IC) realization due to its ease of high‐accuracy realization because the coefficients of the transfer function can be realized with the capacitance ratio of the capacitors. However, when the coefficients of the transfer function are widely spread, the spread of the capacitance is also large, Hence, the total capacitance is eventually increased so that IC realization is difficult.In this paper, a new SC filter configuration is proposed in which the coefficients of the transfer function of the SC transversal filter is rearranged in the order of magnitude and the ratio of the neighboring coefficients is used as the normalized capacitance (ratio to the capacitance of the operational amplifier for coefficient multiplication) of the coefficient capacitors. According to the proposed configuration, the spreading of the capacitance ratio and the total capacitance can be reduced significantly. Since the coefficient ratio of the transfer function is used, there is concern that the characteristic degradation by the element deviations might be larger than the one in the conventional SC transversal filter.It is shown by the element sensitivity analysis and the numerical example that the characteristic variation of the cascaded SC filter by the present configuration is smaller than the one in the conventional cascaded SC filter in which the basic block is made of SC transversal filters and, therefore, there is no problem in practice. Finally, a test result is presented for an SC transversal filter using a third‐order transfer function coefficient ratio.
Design and implementation of linear‐phase two‐channel perfect reconstruction finite impulse response filter banks with equiripple stopband
Authors: Nagai Takayuki , Ikehara Masaaki 
Published: July 2007

AbstractThis paper presents a design and implementation method for the linear phase two‐channel perfect reconstruction finite impulse response (FIR) filter banks with arbitrary filter lengths. This method is labeled a weighted Lagrange‐Newton method; it is obtained by introducing the least squares weighting function into the Lagrange‐Newton method. Filter banks with good stopband attenuation (equiripple stopband) can be designed using this method. Furthermore, a new method is presented that makes the filter banks with a lattice structure using the property of a polyphase component matrix. the filter banks can achieve perfect reconstruction in spite of coefficient quantization.
Tracking system for spread spectrum communication on power line
Authors: Kimura Naoki , Miyashita Teruo , Sato Chikara , Suzuki Rie 
Published: July 2007

AbstractIn the power line data transmission using the direct sequence/spread spectrum (DS/SS) communication system, the frequency band is limited to 10 to 450 kHz. Then, it is useful to apply the Manchester coding so that the spectrum of the pseudonoise (PN) waveform contains a larer power in the given frequency band. As the delay‐lock loop (DLL) for the synchronization tracking of the Manchester‐coded PN waeform, M2/3Δ and M1.5Δ types are known. the M2/3Δ type has the feature that the tracking error is small, and the M1.5Δ type has the feature that the tacking range is wide.This paper proposes the M2/3Δ improved DLL as an improved DLL which has both the excellent trakcing performance of the M2/3Δ type and the wide tracking range of the M1.5Δ type. It is a DLL combining the M2/3Δ and M1.5Δ types the basic equation and the tracking trajectory by a computer simulation are presented. Then the tracking performance is analyzed theoreitcally in the linear interval of the phase comparator characteristics, and the result is compared to the performance of the M2/3Δ type.
Burst error correcting 2k‐ary cyclic an codes generated by natural number A1·p
Authors: Murakami Ryutaro , Shimada Ryosaku , Ohkura Yoshiteru 
Published: July 2007

AbstractCyclic an codes are considered to be suitable for error detection and error correction in arithmetic operations. Cyclic an codes using radix 2k representation are called 2k‐ary cyclic an codes. This paper is concerned with 2k‐ary cyclic an codes for burst error correction. After describing the structure of these codes and arithmetic burst errors, the smallest code generator A1 that provides a 2k‐ary cyclic an code with the burst error correcting ability b is presented. We also suggest burst error correcting 2k‐ary cyclic an codes generated by A1.p. Furthermore, error‐trapping decoding for 2k‐ary cyclic an codes is shown.
Transient analysis of multisection transmission lines using a modified central difference method
Authors: Murakami Kazuhito , Ishii Junya 
Published: July 2007

AbstractIn designing high‐speed pulse integrated circuits, it is necessary to clarify that the propagating waveform is affected by the multiple reflection caused by discontinuity and mismatching of the interconnected lines. In this paper, we present the modified central difference method with simple boundary treatments for solving Telegrapher's equations only in the time domain in which both voltage and current solutions are derived at the same time. As this algorithm has a compact structure and memory independent of time as compared with other algorithms, the transient phenomena including multiple reflection along the whole line can be represented dynamically. It is also available for the steady state. It is shown by simulation studies that inhigh‐speed pulse transmission on microwave analog circuits and digital VLSIs, this technique is simple and useful in analyzing any type of propagating signals on uniform and various nonuniform transmission lines.
Realization of one‐dimensional nonuniform‐band maximally decimated filter banks using two‐dimensional filters
Authors: Fukuoka Takashi , Ozawa Hiroyuki , Ikehara Masaaki , Takahashi Shin‐Ichi 
Published: July 2007

AbstractIn the design of nonuniform‐band filter banks (NFB), each filter must be designed to satisfy the perfect reconstruction (PR) condition corresponding to division ratio for the system. A simple design method for nonuniform‐band maximally decimated filter banks using two‐dimensional (2‐D) filter to realize two‐channel system is presented here. A 2‐D filter is designed using a McCl‐ellan transform and one axis is fixed to arbitrary frequency in a 2‐D plane. Then the frequency response that turns out to be a cross sectional response of 2‐D filter, 1‐D variable filter that has fixed frequency as a parameter can be realized. Using 2‐D filters with complementary power, the lowpass filter and high‐pass filter always satisfies power complementarily. the 2‐D filter has been designed so that the filter pair satisfies the perfect reconstruction condition and it is adopted as the group of the necessary filters for NFB. In this method, the PR condition is not satisfied strictly but system simplification is achieved. Design examples, amplitude responses of them, and simulation results of the reconstruction of the signal are shown.
Tracking capability and floating‐point error analysis in multirate complex recursive weighted least squares algorithm
Authors: Shimizu Jun'ya , Miyanaga Yoshikazu , Tochinai Koji 
Published: July 2007

AbstractThis paper presents an analysis of a multirate complex recursive weighted least squares (MC‐RLS) algorithm based on an analytic signal. Conventional adaptive filters for multiple sinusoid extraction have been based on lattice filter structures or gradient methods because of their low computational cost and/or low sensitivity to quantization errors. On the other hand, the RLS algorithm is easy to introduce assuming time‐variant quantities in the algorithm when sinusoid frequencies have the time‐varying property. However, the relationship between the tracking capability and the quantization error of the least‐squares (LS) algorithm in the transversal structure has not been reported. In addition, an improvement algorithm for these errors have not been reported. In this paper, we shall describe a new RLS algorithm in a transversal filter structure with a superior transient property, reduced sensitivity to quantization errors, and low computational cost. First, an analytic signal‐based autoregressive model is introduced and the MC‐RLS algorithm is shown. Then, using an excess mean‐square error of the MC‐RLS algorithm, the tracking capability shown to be unaffected by the analytic transform and the decimation. In addition, the floatingpoint error and the computational cost of the MC‐RLS algorithm are analyzed. It is shown that both floatingpoint error and computational cost are smaller than those of conventional RLS algorithms that use real signals.
Probabilistic analysis of the time complexity of quicksort
Authors: Mizoi Tadashi , Osaki Shunji 
Published: July 2007

AbstractQuicksort is a well‐known sorting algorithm based on the divided control. the array to be sorted is divided into two sets as follows. an element in the array is specified, and the set of values larger than the value of that element and the set of values smaller than that value are constructed. Each of those two sets are sorted independently. the procedure is iterated for the divided sets. In other words, the algorithm has a recursive structure. the average time‐complexity of the quicksort (the average number of comparisons) is O(n log n). Depending on the data to be sorted, however, the performance may be deteriorated drastically. In the worst case, the time complexity is O(n2).In this paper, the generating function based on the time complexity of the quicksort is introduced and the mean and the variance of the time complexity is determined analytically using the generating function. Based on the derived mean and the variance, the chi‐square test is applied as to whether or not the distribution of the time complexity can be approximated by the normal distribution. Then, a method is proposed in which the probability R(x) that the time complexity exceeds a certain value x is determined accurately by approximating the distribution of the time complexity of the quicksort by a three‐parameter Weibull distribution. Finally, the selection of the sorting algorithm is discussed.