bluuebunny/crossref_metadata_2025_split huggingface dataset:
A new reference signal for evaluating the quality of speech coded at low bit rates
Authors: Asanuma Naomi , Nagabuchi Hiromi 
Published: July 2007

AbstractThis paper proposes a reference signal for use in evaluating the speech quality of low bit‐rate coded signals. The proposed signal is generated by adding a simulated low bit‐rate coding distortion signal to the original signal. This low bit‐rate coding distortion signal is modeled with both spectrum envelope distortion and additive white noise so that the simulated coding distortion signal is synthesized using both linear predictive coding (LPC) parameters derived from a signal whose spectrum envelope is distorted by spectrum‐shaped noise and by an LPC prediction error signal with added white noise. The simulated‐signal parameters, the amount of spectrum‐shaped noise and white noise, approximate the average characteristics of low bit‐rate coding distortion signals. Experimental results show that the proposed reference signal suppresses variability in evaluation scores and improves their reproducibility in repeated assessment tests.
A numerical calculation for some nonlinear optimal control problems with a terminal constraint using synthetic Riccati transformation and its applications to the manipulator control problem
Authors: Cai Dawei , Yamaura Hiroo , Shidama Yasunari 
Published: July 2007

AbstractThis paper describes a new numerical calculation algorithm for the nonlinear quadratic optimal problem with nonlinear terminal constraint. Synthetic Riccati transformation proposed earlier by the authors is utilized to solve the forementioned problem.The state differential equation is changed into two differential equations with respect to the two substates vector selected from the state vector. Then, based on the maximum principle and Kronecker's product, two differential equations with respect to the adjoint variables can be derived. To solve the forementioned differential equations, synthetic Riccati transformation is introduced to obtain a Riccati differential equation and an accessory differential equation. Thus, we can calculate the initial values of the adjoint variables by solving Riccati equations backward in time. The convergence condition about this algorithm also is provided.As the application of this algorithm, an attempt is made to apply it to an optimal trajectory control problem for a two‐link manipulator. As the result of the simulation, it is confirmed that the algorithm not only has a performance of both fast convergence and high computation accuracy but also is not subject to the selection of the normal parameter for computation.
Speech recognition using phoneme HMM constrained by frame correlation
Authors: Takahashi Satoshi , Matsuoka Tatsuo , Minami Yasuhiro , Shikano Kiyohiro 
Published: July 2007

AbstractOne of the problems with the hidden Markov model (HMM) in performing speech recognition is that the local transition information of the feature vectors is not incorporated into the mechanism of the model and the model is not constrained by transitions of the feature vectors. Thus, the output probability distribution never changes during recognition. Furthermore, all transitions between the vectors that have high probabilities are allowed even if those transitions did not appear in the training data.This paper proposes a bigram‐constrained HMM that uses correlations between two frames to constrain the feature distributions of a speaker‐independent HMM to the region most appropriate for the speaker. Since the output probability of the bigram‐constrained HMM is a conditional probability restricted by the feature vector of the previous frame, the output probability changes dynamically at each frame depending on the feature vector of the previous frame. Constraining the feature distribution makes it possible to reduce the overlapping of feature distributions between different phonemes which improves recognition performance.Previously, we proposed the discrete bigram‐constrained HMM which is based on the combination of a discrete speaker‐independent HMM and the VQ‐code bigram. We showed that it performed better than conventional speaker‐independent HMMs. In this paper, the strategy is extended to the tied‐mixture bigram‐constrained HMM and the continuous bigram‐constrained HMM to obtain better recognition performance. These three types of HMMs are formulated and evaluated by phoneme recognition in continuous speech.
An adaptive control for linear uncertain multiinput‐multioutput systems with robust performance
Authors: Maki Midori , Hagino Kojiro 
Published: July 2007

AbstractThis paper proposes an adaptive controller design method for linear multiinput‐multioutput systems with unknown parameters, an accurate model of which is not available. The proposed adaptive control scheme results in the prevention of system performance deterioration caused by the parameter uncertainties. In the proposed method, it is not necessary to estimate unknown parameters and/or to adjust the controller parameters.First, a candidate for an approximate model to describe the uncertain parts of the plant is introduced and a target model which the controlled system should track is constructed based on the known nominal system and this approximate signal. Next, this tracking problem is changed to the linear quadratic (LQ) control one for the augmented system. This target trajectory is determined automatically to be close to the optimal response for the nominal system. It is shown that the asymptotic stability of the resulting total system is ensured if certain stability conditions are satisfied. In addition, an observer‐based adaptive control method and the stability of the combined plant‐observer‐controller system also are discussed when not all the states can be measured.
A realization method for an ARMA lattice filter with reduced number of coefficients
Authors: Haseyama Miki , Nagai Nobuo , Miki Nobuhiro 
Published: July 2007

AbstractThis paper proposes realization methods of two kinds of low sensitivity ARMA lattice filters which can perform frequency‐weighting model identification with input estimation and low sensitivity of their filters. Using the proposed realization method, the filter with the fewer coefficients than the previous filter can be obtained. Since the calculation costs for the realization of the filter are reduced, the proposed realization methods are significant. Furthermore, assuming that the ARMA lattice filters have low sensitivity, however, this paper shows that the proposed ARMA lattice filters have low sensitivity by simulations, which is important for practical implementation of filters.
Resonance frequency analysis by galerkin's method for clamped‐free, stepped flexural mode quartz crystal resonators and its application to a quartz crystal tuning fork
Authors: Kawashima Hirofumi , Sunaga Kenji 
Published: July 2007

AbstractThis paper describes a stepped flexural mode quartz crystal resonator. The objective of this paper is to determine the resonant frequency, the motional series resistanceR1, and theQvalue of the stepped flexural mode tuning fork‐type quartz crystal resonator. The equation of motion is presented for a flexural mode resonator with nonuniform cross section with rotary inertia, and shear force is taken into consideration. Then, since the flexural mode quartz crystal resonator can be thought of as having two clamped‐free beams, the vibrational displacement computed from the boundary condition of clamped‐free ends and the condition of continuity of different cross sections are used to derive the frequency equation by Galerkin's method.In addition, the transcendental equation can be derived from the boundary conditions and the continuity condition. Further, from these relationships, the frequency constant (f·l1) and the resonant frequency are determined as functions of the width ratioW21(widthw2/‐widthw1) and the length ratioL21(lengthl2/lengthl1). As a result, it is shown that the frequency can be lowered without increasing the length of the resonator. These relationships are compared with the experimental results. It is shown that both agree very well. Hence, the present analysis results are effective for arbitrary frequency design. Finally, the typical values of the electrical equivalent circuit parameters are given. It is shown that, asL21is increased, the motional series resistanceR1increases and theQvalue decreases, although the characteristics are sufficient as a resonator. At the same time, the structure is resistant to vibration.
Software operation system information model
Authors: Kobayashi Toru , Horikawa Keitaro , Ushida Fumiaki 
Published: July 2007

AbstractA domain model for software maintenance and delivery is proposed and a specific information model for a telephone switching support system is derived from it by using object‐oriented analysis. This domain modeling and information‐model construction method can easily define boundaries for complicated tasks and can be reused for other target systems. Since the derived information model does not have any redundant objects or services, it is believed that it will facilitate the development of systems that have consistency among objects and specialized services.
Adaptive phase control using recursive least squares (RLS) phase estimation
Authors: Denno Satoshi , Saito Yoichi 
Published: July 2007

AbstractThis paper considers the realization of high‐speed phase synchronization when there exists a frequency error and discusses the application of the recursive least square (RLS) algorithm to the burst signal demodulator. Especially, as a phase control algorithm for the coherent detection where there is an error in the carrier frequency, a method is proposed to represent the phase variation by the frequency error as the state transition function and a detailed algorithm is derived.It is shown that the theoretical convergence and the tracking performance of the algorithm agree with those of the ordinary RLS algorithm. The algorithm is applied to the frequency control circuit of the π/4‐QPSK differential detector and the phase control circuit of the 16‐QAM quasi‐coherent detector.It is demonstrated experimentally that the system has an excellent performance and that the initial convergence is complete in approximately ten symbols when there exists a frequency error.
Weak radar signal detection based on wavelet transform
Authors: Ehara Naoki , Sasase Iwao , Mori Shinsaku 
Published: July 2007

AbstractWeak signal detection is a basic and important problem in radar systems. The detectable radar range is determined by the radar equation, and radar performance can be improved by increasing the receiver output signal‐to‐noise ratio (SNR). The pulse integration method has been used to improve SNR. The new method proposed here, based on wavelet transform, has been used primarily for data compression in image processing. Since wavelet transform performs signal analysis using the wavelets that are localized in both time and frequency domains, better results are expected than would be expected if the conventional method of pulse integration in time domain is used. Using numerical simulations that assume white Gaussian noise is generated internally in the receiver, we show that an improvement of approximately 8 dB in SNR is obtained for only one pulse and high performance radar can be obtained using this method. This method can also be applied for signal detection in other digital communications.
Speaker adaptation using spectral interpolation for speech recognition
Authors: Shinoda Koichi , Iso Ken‐Ichi , Watanabe Takao 
Published: July 2007

AbstractThis paper proposes a technique for supervised speaker adaptation for speech recognition based on a continuous density hidden Markov model (HMM). When the number of utterances for the adaptation words is decreased in order to reduce the burden on the user, the ratio of demisyllables (recognition units) not appearing in the uttered words is increased. In the proposed method, the HMM parameters for the demisyllables contained in the adaptation data are corrected and then the HMM parameters for the demisyllables not contained in the adaptation data are corrected by interpolation in the parameter space (spectral interpolation). To avoid a biased estimation of the parameters that depend on the adaptation data set, a correction is made based on large‐scale uttered data by a large number of speakers. A word recognition experiment, executed for 100 similar words, simulates a 5000‐word large‐vocabulary speech recognition; and the proposed method is then evaluated. When the recognition rate by the speaker‐independent HMM reaches 81.2 percent, the recognition rate is improved up to 85.2 percent by using 50 words for adaptation. Thus, the effectiveness of the proposed method is verified.
Vector quantization of LSP parameters using moving average interframe prediction
Authors: Ohmuro Hitoshi , Moriya Takehiro , Mano Kazunori , Miki Satoshi 
Published: July 2007

AbstractThis paper proposes a new method for vector quantization (VQ) of LSP parameters, using moving average (MA) interframe predictions. MA predictive VQ executes the prediction using the code vectors (codebook outputs) in the previous frames. Moreover, this method has the following features, compared to autoregressive (AR) predictive coding, which executes the prediction based on the previous quantized (decoded) values: (1) even if a bit error is produced in the transmission channel, its effect to the succeeding frames remains finite; and (2) the stored codes can be decoded from any time point.This paper discusses the quantization performance and the robustness against bit errors of the MA predictive VQ, as well as the training method for the codebook. Assuming an application to low‐bit‐rate speech coding, the configuration considered has a frame length of 40 ms, with each frame composed of four subframes. A method for further improvement of the efficiency is reported. As a result of evaluation using actual speech data, bit reductions of approximately 16 percent and 23 percent are achieved for the quantization for each 20 ms and 40 ms, respectively, compared to the conventional method which does not use interframe prediction.
Fuzzy programming for large‐scale multiobjective linear programming problems with block angular structure
Authors: Sakawa Masatoshi , Inuiguchi Masahiro , Sawada Kazuya 
Published: July 2007

AbstractSince Dantzig and Wolfe proposed the decomposition principle for the large‐scale linear programming problem with the block angular structure in the early 1960's, there have been made a large number of intensive studies for the large‐scale mathematical programming problems with special structures. Considering the fuzziness of the human decision, however, it seems more adequate to assume that the decisionmaker has the fuzzy goals or fuzzy constraints for the objective functions and for the coupled constraints in the problem formulated as the large‐scale linear programming problem with the block angular structure.From such a viewpoint, this paper focuses on the large‐scale multiobjective linear programming problem and proposes a new method to derive easily the satisficing solution for the decisionmaker considering his/her fuzzy goal. In the proposed method, the fuzzy goal of the decisionmaker for each objective function is specified by the linear membership function through the interaction with the decisionmaker. Fuzzy decision is applied to integrate the fuzzy goals. Then the original problem is decomposed into a master problem and several independent small‐scale linear programming subproblems.The Dantzig‐Wolfe decomposition principle is applied, where the optimal solution as a whole is obtained through iterations, by updating the goal function of each subproblem according to the solution of the master problem. Thus, it is shown that the satisficing solution is obtained relatively easily considering the fuzzy goals of the decisionmaker.
Analysis of modified transmission line antenna for a portable telephone
Authors: Kumon Yasunori , Tsukiji Takehiko 
Published: July 2007

AbstractIn previous works, the authors have developed a mobile communications antenna and a high‐frequency communications antenna using the modified transmission line antenna with good results. The modified transmission line antenna is a kind of low‐profile transmission line antenna of half a wavelength. It has the advantage that desirable operating characteristics can be chosen in the antenna design because it is relatively easy to control impedance and gain by modifying its shape.In this paper, for the purpose of developing an antenna for portable telephones, a modified transmission line antenna is proposed for a portable telephone with this antenna mounted on a conducting box representing the body of the portable telephone. This configuration was analyzed using the wire‐grid method. As a result, impedance, gain, and directivity characteristics which make this antenna suitable for portable telephones were obtained and the results are described here. Furthermore, if the conducting body representing the telephone is 100 to 160 mm long, it has a relatively small effect on the antenna characteristics, which is useful for miniaturization of the telephone body. From the calculated results and measurement results, the effectiveness of the wire grid analysis method is verified.Finally, the case is analyzed where the modified transmission line antenna is mounted on a folding portable telephone body, which is becoming popular. It is shown that the folding angle of the telephone does not significantly affect its characteristics.
Pitch synchronous innovation code excited linear prediction (PSI‐CELP)
Authors: Miki Satoshi , Moriya Takehiro , Mano Kazunori , Ohmuro Hitoshi 
Published: July 2007

AbstractThis paper proposes a new speech coding method pitch synchronous innovation code excited linear predictor (PSI‐CELP). This method is based on CELP but adds pitch synchronous innovation. This results in even random codevectors being adaptively converted to have pitch periodicity for voiced frames. This scheme can improve the synthesized speech quality of voiced frames in the low bit‐rate CELP without increasing either computational complexity or bit rate.In addition to pitch synchronous innovation, this paper also proposes four other methods in which the quality of synthesized speech is effectively improved or the computational complexity is reduced for low bit‐rate CELP: (1) the perceptual weighting filter is an MA‐type and uses nonquantized LPC parameters; (2) the fixed codebook is prepared as part of the adaptive codebook for nonperiodic frames; (3) the random codebook consists of two‐channel conjugate sub‐codebooks; and (4) codebook searches are made using delayed decision. In a speech coding system with pitch synchronization of random codevectors, improvements in ratio SNR, segmental SNR, and opinion equivalent Q value are respectively 1.6 dB, 1.0 dB, and 0.5 dB. These basic methods were adopted in the standard codec for Japanese half‐rate digital mobile telephone service.
Noise cancelling for the digital holter monitoring system
Authors: Kohama Takashi , Nakamura Shogo 
Published: July 2007

AbstractThe present trend is to develop a totally digital Holter electrocardiogram (ECG) recording system. To this end, research and development are in progress for the digital Holter ECG as the recorder. The conventional Holter ECG is a device which is carried for long hours and records the ECG waveforms on magnetic tape as the analog signal. It has problems in accuracy, e.g., waveform distortion, moving parts such as the motor, as well as volume and weight.The digital Holter ECG considered in this paper is the simplified digital Holter ECG which converts ECG waveforms for 24 hours into digital form, to record and process various waveforms. By totally designing the system in a digital form, the circuit can easily be integrated into LSI. By using the IC memory card as the recording medium, a small‐sized and lightweight equipment can be realized. Thus, the problems in the past system can be solved.A problem in the present IC memory card is that the recording capacity is limited; and the data compression of approximately 1/60 is required to record ECG waveforms for 24 hours. The achievable compression ratio is affected by the noise on the ECG waveform and the fluctuation of the baseline.This study is concerned with the method to eliminate the noise, which is a problem in the data compression. The results of analysis and simulation are presented.
Generalization of smoothness measure for standard regularization and its application to surface reconstruction
Authors: Mizutani Hiroyuki 
Published: July 2007

AbstractThe standard regularization is a theory that converts ill‐posed problems into well‐posed problems by assuming some a priori information about solutions. Usually, this assumption is concerned with the smoothness of a function that is a solution of the problem. To measure the smoothness of a function, a membrane or thin‐plate measures have been used, i.e., the deformation energy of a membrane and a thin‐plate, respectively. The smoothness measure is generalized by using a continuous order differential‐integral operator. This generalized measure is applied to surface reconstruction from sparse data and its usefulness and superiority to these measures are shown.
Problem of computing the expected maximum number of vertex‐disjoint s‐t paths on probabilistic digraphs
Authors: Cheng Peng , Masuyama Shigeru 
Published: July 2007

AbstractThe problem of finding vertex disjoint paths (i.e., paths sharing no vertices) between a specified pair of vertices of a digraph has been extensively studied as one of the fundamental problems in graph theory. For digraphs whose vertices or edges have mutually independent failure probabilities (called probabilistic digraphs hereafter), the expected maximum number of vertex disjoint paths is a measure of reliability of the network and its computation is an important practical problem. In this paper, the problem of computing the expected maximum number of vertex disjoint paths between a specified pair of vertices in a given probabilistic digraph is considered. It is shown that the problem is in general NP‐hard but that the expected value can be obtained in polynomial time for special types of probabilistic digraphs such as probabilistic s‐t series‐parallel digraphs and probabilistic s‐t complete multistage digraphs in which the vertices have the same failure probability and the arcs are failure‐free. Further, the problem is shown to be NP‐hard for the following three cases: probabilistic planar digraphs; probabilistic s‐t out‐in bitrees; and probabilistic s‐t complete multistage digraphs.