bluuebunny/crossref_metadata_2025_split huggingface dataset:
Automatic learning in chaotic neural networks
Authors: Watanabe Masataka , Aihara Kazuyuki , Kondo Shunsuke 
Published: July 2007

AbstractA fully local algorithm which can automatically detect and learn an unknown pattern is proposed for a mutually connected recurrent neural network, and its fundamental properties are numerically analyzed. the algorithm is applied to chaotic neural networks composed of neuron models with spatiotemporal inputs and refractoriness and to conventional mutually connected neural networks. It is shown that the former could learn more patterns with greater robustness than the latter.
Use of static/dynamic parameters in automatic phonemic segmentation system for english continuous speech
Authors: Furuichi Chieko , Aizawa Katsura , Imai Satoshi 
Published: July 2007

AbstractIn this paper, a phonemic segmentation system is constructed which will be useful in practice as a preprocessing for English continuous speech recognition. the usefulness of the system is demonstrated. English continuous speech contains many weakly voiced vowels with ambiguous utterances called schwa, making stable detection of the phonemic boundary difficult. In fact, there has been almost no detailed proposal for a segmentation system that can be used in practice as a preprocessing to the recognition.In the proposed method, the mel cepstrum is extracted from the speech signal, using the unbiased estimation of the log spectrum. This process is known to be stable, i.e., less affected by the fine spectral structure. Then the dynamic segmentation parameters are determined from the mel cepstrum such that the boundary of the schwa phonemes can be detected using the pseudo‐differentiation filter. By combining the dynamic parameters with the static parameters, the phonemewise segmentation is executed in a hierarchical form.The proposed system can segment English continuous speech containing diversified phonemic environments into phonemes on the time axis. the system operates based only on the acoustic knowledge of phonemes, which are common to speakers with different qualities and utterances, without using the speaker‐dependent complex boundary detection rules. Also, the proposed system is evaluated by experiment using English continuous phoneme‐balanced speech uttered by one each native English‐speaking male and female for 350 s. For the total number of phonemes 3024, the detect ratio for the phoneme boundary is 97.1 percent, the boundary delete ratio is 2.9 percent, and the boundary add ratio is 24.2 percent.
Two‐dimensional spectral estimation method with data extension and its improvement
Authors: Shimamura Tetsuya , Miao Weiguo , Suzuki Jouji 
Published: July 2007

AbstractThis paper considers the application of the two‐dimensional (2‐D) spectral estimation for estimating the frequency of the 2‐D sinusoidal signal from a small number of data arrays. Comparing the 2‐D autocorrelation method, which is a typical 2‐D method of linear prediction, and the data extension method proposed by Frost et al., it is shown that the data extension method gives a better result of spectral estimation. It is noted that the estimation accuracy is deteriorated when an additive noise is mixed.From such a viewpoint, this paper considers first the noise‐compensation method, where the noise power is subtracted from the diagonal elements of the correlation matrix in the Yule‐Walker equation in the one‐dimensional (1‐D) processing. Then, a simple and efficient noise‐compensated autoregression (AR) parameter estimation method is derived, which can solve at the same time the problems of the accurate estimation of the noise power and the deterioration of the AR parameter estimation accuracy due to the oversubtraction of the estimated noise power, whch have been considered unsolved.Applying the idea to the data extension method, an improved data extension method is derived. It is shown that by this improvement, the accuracy of the data prediction is improved and the estimation of the 2‐D spectrum is improved greatly.
An identification method for nonminimum‐phase autoregressive moving average system using phase‐equivalent moving average system
Authors: Miyazaki Nozomi , Hamada Nozomu 
Published: July 2007

AbstractOne of the most important issues of signal processing is to identify the model that can describe the generation process for an observed signal through the statistical analysis of the random signal. Recently, as an identification method for such process, a method is considered in which the phase information of the signal using higher‐order cumulants is paid close attention. This paper proposes a method in which the observed one‐dimensinal signal is modeled as the output signal of a linear time‐invariant system generated with the non‐Gaussian white process. the generation systm for the signal (not constrained to be minimum phase) is identified using the third‐order cumulant of the observed signal. Because the system characteristics are decomposed into their components in the cepstral domain, the proposed method does not require the factorization procedure. In other words, the feature of the method is that it is extendable to the multidimensional system identification, for which the factorization generally is impossible.
On a decoding algorithm by superposition for binary cyclic codes for certain burst errors
Authors: Ogino Atsushi , Kohnosu Toshiyuki , Hirasawa Shigeichi , Nishijima Toshihisa 
Published: July 2007

AbstractCyclic codes are often used for the purpose of correcting burst errors in digital transmission and recording systems. Since the implementations of the encoding and syndrome calculations of these codes are simple, these codes have superior properties from a practical perspective. In this paper, an effective decoding method is given for certain burst errors (multiple solid‐burst errors) that are contained within the detected errors resulting from a binary cyclic “predecoding” proces, where the precoding process for burst error correction is terminated at the detection stage. Such burst errors can be expressed as random errors by superimposing a copy of itself that has been cyclically shifted by one bit. So, by performing random error decoding on a version of the received sequence that has been superimposed with a one‐bit cyclically shifted copy of itself, burst errors can be accurately corrected. This decoding method completely preserves the correction capacity of the precode and is applicable to any binary cyclic code with a minimum distance of at least 5. the correctable multiple solid‐burst errors depend on the minimum distance of the code. Furthermore, by applying this technique to interleaved random error‐correction codes for the purpose of correcting burst errors, it is shown that it is possible to extend the length of correctable burst errors.
Double‐talk detection method with detecting echo path fluctuation
Authors: Fujii Kensaku , Ohga Juro 
Published: July 2007

AbstractIn the echo canceller, where the normalized least mean square (NLMS) algorithm is used as the adaptive algorithm, the double‐talk disturbs the coefficients of the adaptive filter and increases the probability of howling generation. to cope with this phenomenon, the echo canceller usually is provided with a detector to detect the generation of the double‐talk. the echo canceller is designed to minimize the disturbance by stopping the update of the adaptive filter coefficients when the double‐talk is detected. In the detection of the double‐talk, it is effective to monitor the increase of the residual echo since the double‐talk component masked by the echo can be detected. In this method, however, the residual echo also is increased by the fluctuatio of the echo path and the fluctuation of the echo path should also be detected.This paper proposes a method to differentiate between the double‐talk and the echo path fluctuation, based on the magnitude of the correlation between the input‐containing echo and the echo replica. In other words, the property is utilized that the correlation between the echo replica and the echo is large for the double‐talk and is small for the echo path fluctuation. A double‐talk detection method also is proposed where the power ratio of the residual echo and the echo replica is used as the parameter. It is shown that the double‐talk detection without deteriorating at all the echo elimination can be realized by a structure with a delay preceding the coefficient update operation.
A new approach to global routing for three‐metal‐layer gate arrays
Authors: Terai Masayuki , Sato Koji , Takahashi Kazuhiro , Matsuzaki Hideo 
Published: July 2007

AbstractWe propose a new global routing approach for three‐metal‐layer gate arrays. A gate array chip handled in this paper consists of rows of cells of a uniform height, megacells such as RAMs or ROMs, and routing channels. The proposed approach breaks the gate array chip into a two‐dimensional array of rectangular global routing cells (in short, G cells) by coarse grid lines and constructs global routes for all the nets on the array of G cells. The feature of our approach is that the capacities of G‐cell boundaries corresponding to routing channels are variable. Utilizing freedom in the capacities of G‐cell boundaries, our global routing algorithm changes the capacities at a certain stage of its execution and reduces the number of nets rerouted in the subsequent rerouting phase, in order to speed up the global routing process. The effectiveness of our approach is demonstrated by our experimental results on 400 K‐gate sea‐of‐gates array chips.
Automatic generation of command level simulation model of a processor from rt level description
Authors: Akaboshi Hiroki , Yasuura Hiroto 
Published: July 2007

AbstractIn designing a digital circuit with a microprocessor, there are several simulation levels. Using a logic synthesis tool and a layout synthesis tool, a lower‐level simulation model can be generated from a design RT level described in HDLs. There is trade‐off between accuracy and speed of simulations. A lower‐level simulation model can simulate more accurately. There is little support for generating high‐level simulation models in which a human designer is required. In this paper, an algorithm is proposed for high‐level simulation model generation and experimental results are shown.
Modeling and simulation analysis of indoor packet radio communication systems
Authors: Kasahara Hirokazu , Hara Shinsuke , Morinaga Norihiko 
Published: July 2007

AbstractThe characteristics of wireless channels are greatly different from those of wire channels. In wireless channels, the capture effect and hidden terminal problems occur. Therefore, it is important that the characteristics of wireless channels are modeled accurately and efficiently when analyzing the total system performance of wireless systems. In this paper, we model the spatial distribution of mobile terminals and base stations for an indoor packet radio communication system. We also model a radio propagation path in terms of Rayleigh fading, logarithmic normal shadowing, and propagation path loss. We then evaluate the packet reception success rate from the channel use, condition of the propagation path, the given modulation/demodulation method and media access protocol, using a bit error rate table. We show a method of analyzing the system characteristics (through‐put, packet success rate, and average delay time) efficiently using computer simulations. Also, as an example of numerical analysis, we create simulation for an indoor radio packet communication system using the direct sequence/spread spectrum (DS/SS) method and analyze its system characteristics.
Error detection of real‐number input fast fourier transform networks
Authors: Chen Song , Goto Munehiro 
Published: July 2007

AbstractFast Fourier transform (FFT) is one of the techniques indispensable in signal processing. the recent development of hardware and parallel processing techniques makes it possible to use a specific chip circuit for signal processing such as FFT and in fact there exist systems that achieve high‐speed computation of FFT. Such systems use many processors in a chip. Consequently, these are likely to have high error probability and require on‐line fault detection. This paper treats the fault‐detection methods of the FFT‐CN (FFT calculation network) under the assumptions that the FFT‐CN is made up ofN/2, x log2Ntwo‐input butterfly modules, that itsNinputs are all restricted to the real values, and that the detection of the fault is performed by the comparison of the outputs of the network. the assumed structure of the FFT‐CN is symmetric so a simple fault‐detection method would not be successful. But we will show here that the restriction of the inputs to real numbers makes it possible to find a fault in the FFT‐CN.
Characteristics of short‐term memory of tone duration in the human auditory system
Authors: Kanoh Shin'ichiro , Futami Ryoko , Hoshimiya Nozomu 
Published: July 2007

AbstractInformation for “tone duration” (duration information) is an important parameter in representing the temporal sequence. This paper reports on the results of a basic auditory psychophysical experiment which was executed in order to estimate how information is represented in the process to be perceived and retained in the brain. the experiment focuses on the “gradual change of the retained duration information,” which has been intentionally excluded in the past acoustic psychology, in order to focus on the perception process. the subject is instructed to compare the durations of two successively presented tones. the experiment demonstrated that the interval of uncertainty, which is the index of forgetting the retained information, depends on the duration of the first tone as well as the interstimulus interval between the two tones. It is then suggested that there exists a “duration which is especially easily forgotten.” A result is also obtained that is consistent with the conjecture that the presented duration information is handled almost as a continuous value in the brain. It is expected that those findings will provide a clue to the modeling of temporal sequence processing in the brain.
The minimum number of sliding operations of palettes in a two‐dimensional automatic warehouse of size N × M (N, M ≥ 3) with three spaces
Authors: Kawaguchi Kimio , Wu Jingjun , Wada Koichi , Aoki Hitoshi 
Published: July 2007

AbstractTo utilize fully a planar automatic warehouse with a specified area, a larger number of goods should be stored and shipped efficiently. Consider the situation in which a planar rectangular automatic warehouse has N × M sections and (NM‐k) palettes 1 × 1 in size. Then, k sections are produced which are spaces without a palette. In the shipping operation process, the palette already loaded with the desired good at the position (n, m) (0 ⩽ n ⩽ N −1, 0 ⩽ m ⩽ M −1) slides to the exit at position (0, 0) by utilizing the places where the good is unloaded. Considering the N × M automatic warehouse with k spaces, a problem is posed to determine the function T(k; n, m), which represents the minimum time (the minimum number of parallel sliding operations, called minimum steps) needed for a palette at position (n, m) to slide to the position (0, 0); T(1; n, m), T(2; n, m), and T(k; n, m) (k4) are already derived by past studies [1, 2], but the derivation of T(3; n, m) has not been achieved. the purpose of this paper is to present the solution for this problem for the case of N ≥ 3, M ≥ 3.
Estimation of stochastic model for random moving image by means of noncausal model
Authors: Miyagi Shigeyuki , Ogura Hisanao , Takahashi Nobuyuki 
Published: July 2007

AbstractA new method for estimating a stochastic model of a random moving image is proposed. Assuming that the spatiotemporal spectrum of a moving image is separable in space and time, a stochastic image model is obtained by combining a noncausal model in two‐dimensional space and a causal AR model in time. Model estimation uses the method of minimizing whiteness for the noncausal spatial model and the method of “sequential orthogonalization” for the causal temporal model. the separable spectrum can be readily applied to a drifting moving model. Applying the spatial filter to a drifting moving image, the moving image is decomposed into a set of uncorrelated time series along the drifting time axis; this allows the AR models to be estimated separately. It is confirmed by computer simulation that the correct model can be estimated by this method. On the other hand, in the model estimation for a real, natural moving image, spectral separability is not always valid and, therefore, the output time series of the spatial filter is not rigorously uncorrelated. For such a real moving image, the method devised is an extended AR model which takes into account the spatially correlated data for AR model estimation. For estimating an extended AR model, the method of quasisequential orthogonalization is developed by modifying the sequential orthogonalization. It is successfully applied to processing a real moving image taken from a video recording.
A second‐order active bandpass filter with complex coefficients and its applications to the hilbert transform
Authors: Zhang Xiaoxing , Shinada Yuji 
Published: July 2007

AbstractIn this paper, the direct realization of a second‐order active bandpass filter with complex coefficients using operational amplifiers, resistors, and capacitors is proposed and its application to the Hilbert transform is discussed. Then, an active analytic signal converter is realized through real coefficient circuits that are required for signal processing with complex coefficients. Finally, examples of these realizations are presented. the feasibility of the proposed circuit is confirmed by the simulation results which are in good agreement with the theoretical values.
Dependability assessment methods for factory automation systems using safety index
Authors: Mineo Yoshiyuki , Suzuki Yoshihisa 
Published: July 2007

AbstractTo evaluate the dependability of systems from the viewpoint of human safety, the magnitude of accidents in addition to the probability of the occurrence of accidents must be taken into consideration. Also, it is essential to be able to cope with various levels of accidents consistently, not to set different criteria according to the system.Research is underway to standardize the safety index and put it to practical use as a criterion to quantitatively decide the dependability of the systems from the viewpoint of safety for people. However, as the safety index had the defect of being unable to represent the magnitude of accidents, its use as an index for evaluation was limited. to overcome this problem, this paper shows that the tree analysis for safety (TAS), which is a combination of the fault‐tree analysis (FTA) and human effectiveness, can be classified by factors of physical accidents such as high‐velocity machinery or high‐temperature substances. A method of evaluating such safety indices according to the criteria classified by the potential of the factors of physical accidents is proposed. By applying this method to examples of factory automation (FA) system accidents, whose patterns are seven at normal operation and four at maintenance, it is effective as a practical system evaluation method.
Stability‐guaranteed adaptive algorithm and its application to active noise canceling
Authors: Ojiro Naotaka , Kajiki Makoto , Ohmori Hiromitsu , Sano Akira , Tsujii Hiroyuki 
Published: July 2007

AbstractThere are some cases where feedforward control can be utilized effectively, e.g., the active noise control and the vibration control of a building. When there exists an indeterministic factor or a fluctuation, the adaptive filter must be introduced. the algorithm that adjusts the coefficient of the adaptive filter has mostly been based on the filtered‐x algorithm. A problem in this approach, however, is that the stability is not guaranteed and the stepwidth of least‐mean square (LMS) algorithm must carefully be determined.From such a viewpoint, this paper proposes the adaptive algorithm both in the time and the frequency domains, where an arbitrary stepwidth can be used, and the robust stability is guaranteed even if an additive noise is mixed. As the first step, the infinite impulse response (IIR) adaptive filter is considered, aiming at implementation of the controller in the time domain, and the properties of the adaptive algorithm are analyzed. Then, the adaptive algorithm in the frequency domain is proposed where the controller is implemented using the frequency‐sampling filter and the stability is guaranteed.In contrast to the case of the design in the time domain, the implementation in the frequency domain has the advantage that the realization is ensured independently of the order or the structure. Finally, the usefulness of the proposed adaptive algorithm is demonstrated by numerical example and experiment in the active noise canceler.
Representation and detection of multiscale, multiorientation fields using local differentiation filters
Authors: Shizawa Masahiko , Iso Toshiki 
Published: July 2007

AbstractA basic theory is proposed for detecting multiorientation fields from images that have multiple orientations at each point in the field. This theory, representing a multiorientation field by a single set of fundamental constraint equations, differs from conventional methods which use many filters tuned to different orientations. It is also different from the steerable filters proposed by Freeman and Adelson [12]. This theory renders implementation of a selection process of orientation‐tuned filters with high‐magnitude output unnecessary. It also makes it unnecessary to implement a search process for an extreme value output of steerable filters. This theory lets analytical solutions to be derived that explicitly compute multiple orientations. This theory is derived from the operational formalism of the principle of linear superposition. Conventional methods suffer from interference between component signals when multiorientation signal components fall in the tuning range of each filter. As a result, filters with a narrow tuning range are necessary in conventional methods. the algorithms derived from out theory are not negatively influenced by interference. By using these algorithms in multiscale image representation, characteristic image structures such as junctions can be extracted from low‐order image derivatives. This theory is expected to provide a theoretical foundation for the image representation of multiple orientations of the hypercolumn structure in the primary visual cortex of the brain.
Maximum‐weight‐window problem of a planar graph
Authors: Yamaguchi Kazuaki , Kotani Ken , Masuda Sumio , Kashiwabara Toshinobu 
Published: July 2007

AbstractWhen a planar graph is drawn, the drawing plane is divided into several regions, each of which is called a face. A subgraph composing the periphery of a face is called a window. For each window W that exists in the drawing of a planar graph G, the total sum of the weights of the vertices and the edges is called the weight of the window W. Among all windows that can appear in drawings, the window with the maximum weight is called the maximum‐weight‐window of G.This paper considers the problem of determining the drawing with the maximum‐weight‐window. the result for this problem is already obtained for the case where the graph is 2‐connected, as well as for the case where the graph is not necessarily 2‐connected but all weights are nonnegative. This paper considers also the arbitrary planar graph with real‐valued weights, and shows that the drawing with the maximum‐weight‐window can be determined in O(n + e) time. This result can be applied to the layout design for LSI and printed board, as well as to the diagram representation of the network. It can also be applied to the solution of the maximal planarization problem for the graph.