Language：English   Publishing type：Research paper (scientific journal)   Publisher：Institute of Electrical and Electronics Engineers (IEEE)  

DOI： 10.1109/taslp.2023.3263776

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Publishing type：Research paper (scientific journal)   Publisher：MDPI AG  

Nonlinear distortion in loudspeaker systems degrades sound quality and must be properly compensated for by linearization techniques. One technique to reduce nonlinear distortion is to use a Volterra Filter, which approximates the nonlinearity of the target loudspeaker using the Volterra series expansion. In general, the Volterra Filter is computationally very expensive, and the amount of computation needs to be reduced for real-time processing. In this paper, we propose an efficient implementation of the third-order Volterra filter based on singular value decomposition. The proposed method determines the necessary coefficients based on the symmetry of the third-order Volterra filter and applies singular value decomposition to them. In the filter structure consisting of singular values and their corresponding singular vector, the computational complexity of the third-order Volterra filter can be reduced by eliminating the part of the filter with small singular values. By focusing on the magnitude of the singular values, the proposed method can improve the computational efficiency of the third-order Volterra filter without decreasing its approximation accuracy. Simulation results show that the proposed method can improve the computational efficiency by 60% while maintaining the nonlinear distortion compensation performance of the micro-speaker for smartphones by about 8 dB.

DOI： 10.3390/app122110710

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Publishing type：Research paper (scientific journal)   Publisher：Institute of Electrical and Electronics Engineers ({IEEE})  

DOI： 10.1109/TCE.2021.3137474

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Publishing type：Research paper (international conference proceedings)  

In this study, we propose a method for the adaptation of a sound source localization model trained on simulation to real-world data in a developed method of a source localization inside a structure. The model for predicting a position of the source is constructed from deep neural network or convolutional neural network, and predicts the source position inside the structure from the frequency spectrum that the accelerometers measure on the outer surface of the structure. The proposed method uses a domain transfer model that transforms real data into pseudo-simulation data to improve the source localization performance of the trained model. The domain transfer model is built from an autoencoder or deep convolutional autoencoder and transfers the data from real to simulation data. The performances of both models is evaluated using the real data as semi-supervised data conditions. A deep convolutional autoencoder led the sound source localization model to a higher than baseline performance.

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Institute of Electronics, Information and Communications Engineers (IEICE)  

DOI： 10.1587/transfun.2021eap1013

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)  

DOI： 10.1121/10.0000515

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：Institute of Electronics, Information and Communications Engineers (IEICE)  

DOI： 10.1587/transfun.e102.a.1932

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)  

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：Elsevier BV  

DOI： 10.1016/j.jsv.2019.04.016

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Institute of Electronics, Information and Communications Engineers (IEICE)  

DOI： 10.1587/transfun.e101.a.2419

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：MDPI AG  

In this paper, we propose a multichannel active noise control (ANC) system with an optimal reference microphone selector based on the time difference of arrival (TDOA). A multichannel feedforward ANC system using upstream reference signals can reduce various noises such as broadband noise by arranging reference microphones close to noise sources. However, the noise reduction performance of an ANC system degrades when the noise environment changes, such as the arrival direction. This is because some reference microphones do not satisfy the causality constraint that the unwanted noise propagates to the control point faster than the anti-noise used to cancel the unwanted noise. To solve this problem, we propose a multichannel ANC system with an optimal reference microphone selector. This selector chooses the reference microphones that satisfy the causality constraint based on the TDOA. Some experimental results demonstrate that the proposed system can choose the optimal reference microphones and effectively reduce unwanted acoustic noise.

DOI： 10.3390/app8112291

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Authorship：Last author   Language：Japanese   Publishing type：Research paper (scientific journal)  

DOI： 10.5687/iscie.31.184

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：Elsevier Ltd  

In this paper, we propose a modified second-order nonlinear infinite impulse response (IIR) filter for equalizing the frequency response and compensating nonlinear distortions of an electrodynamic loudspeaker. A problem of electrodynamic loudspeakers is the generation of nonlinear distortions, which degrade the sound quality. One of the approaches to reducing nonlinear distortions is to use a second-order nonlinear IIR filter. This filter is based on an equivalent circuit model of an electrodynamic loudspeaker and its coefficients are determined by physical parameters. However, it is difficult to compensate nonlinear distortions when a loudspeaker has a high Q factor at the lowest resonance frequency, at which the displacement of the diaphragm and nonlinear distortions become large. The Q factor determines the linear frequency response of the electrodynamic loudspeaker. Although it is necessary to compensate the Q factor of the loudspeaker, a nonlinear IIR filter cannot compensate the Q factor because it does not have a linear filtering feature. In this paper, we propose a modified second-order nonlinear IIR filter that can not only compensate the nonlinear distortions caused by the nonlinearities of the force factor and stiffness but also equalize the frequency response by employing the linear characteristics of the loudspeaker with the desired Q factor. Experimental results show that the proposed filter can compensate the linear and nonlinear distortions more effectively than a conventional filter.

DOI： 10.1016/j.apacoust.2017.11.014

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Authorship：Last author   Language：Japanese   Publishing type：Research paper (scientific journal)   Publisher：Institute of Electrical Engineers of Japan  

We analyze the adaptation rate of active noise control using a statistical-mechanical method. Three measures are employed to evaluate the adaptation rate. The first measure is the MSE initial decreasing rate. The second measure is an adaptation constant, which is defined as the negative of the maximum eigenvalue of the coefficient matrix of differential equations that describe the dynamical behaviors of the macroscopic variables. The third measure is the integral of the MSE, which is defined as the integral of the difference between the MSE and the steady-state MSE. The first and second measures focus on only the initial and final stages of the MSE, respectively. In contrast, the third measure considers all stages in a well-balanced manner. Therefore, employing the integral of the MSE, we can determine the step size that optimizes the entire learning curve.

DOI： 10.1541/ieejeiss.138.369

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：PERGAMON-ELSEVIER SCIENCE LTD  

When estimating the correlation/spectral structure of a locally stationary process, one has to make two important decisions. First, one should choose the so-called estimation bandwidth, inversely proportional to the effective width of the local analysis window, in the way that complies with the degree of signal nonstationarity. Too small bandwidth may result in an excessive estimation bias, while too large bandwidth may cause excessive estimation variance. Second, but equally important, one should choose the appropriate order of the spectral representation of the signal so as to correctly model its resonant structure when the order is too small, the estimated spectrum may not reveal some important signal components (resonances), and when it is too high, it may indicate the presence of some nonexistent components. When the analyzed signal is not stationary, with a possibly time-varying degree of nonstationarity, both the bandwidth and order parameters should be adjusted in an adaptive fashion. The paper presents and compares three approaches allowing for unified treatment of the problem of adaptive bandwidth and order selection for the purpose of identification of nonstationary vector autoregressive processes: the cross-validation approach, the full cross -validation approach, and the approach that incorporates the multivariate version of the generalized Akaike's final prediction error criterion. It is shown that the latter solution yields the best results and, at the same time, is very attractive from the computational viewpoint. (C) 2017 Elsevier Ltd. All rights reserved.

DOI： 10.1016/j.automatica.2017.04.033

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)  

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：PHYSICAL SOC JAPAN  

We analyze the behaviors of the least-mean-square algorithm for a time-varying unknown system using a statistical-mechanical method. Cross-correlations between the elements of a primary path and those of an adaptive filter and autocorrelations of the elements of the adaptive filter are treated as macroscopic variables. We obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under conditions in which the tapped delay line is sufficiently long. We analytically show the existence of an optimal step size. This result is supporting evidence of Widrow et al.' s pioneering work that clarified the trade-off between the noise misadjustment and the lag misadjustment. Furthermore, we obtain the exact solution of the optimal step size in the case of a white reference signal. The derived theory includes the behaviors for a time-constant unknown system as a special case.

DOI： 10.7566/JPSJ.86.024803

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：ELSEVIER SCI LTD  

This paper reports the binaural active noise control (ANC) system developed to deal with factory noise. The control points are located in the vicinity of the left and right ears of a worker sitting along the production line. Due to the complicated safety requirements in the factory, secondary sources and error microphones are not allowed to be placed near the worker. Therefore, the proposed ANC system employs the feedforward structure and adopts the parametric array loudspeakers (PALs) as the secondary sources. The PAL is a type of directional loudspeaker that generates a much narrower sound field as compared to the conventional loudspeaker. Once the proposed ANC system has been trained offline, the error microphones can be removed. The performance of the binaural ANC system is successfully demonstrated based on a digital signal processor (DSP) implementation. (C) 2016 Elsevier Ltd. All rights reserved.

DOI： 10.1016/j.apacoust.2016.09.021

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：ELSEVIER SCI LTD  

The parametric array loudspeaker (PAL) is a type of directional loudspeaker that utilizes the nonlinear acoustic effect to create the audible sound in an ultrasonic beam. Due to this unusual sound principle, it is inevitable that nonlinear distortion is incurred in the sound transmission of the PAL. Numerous modulation methods aiming to reduce the nonlinear distortion have been developed on the basis of the Berktay's far-field solution, but they often perform in an unexpected manner. The degraded practical performance has been credited to the inaccuracy of the Berktay's far-field solution. In this paper, we demonstrate the effect of the ultrasonic emitter on the distortion performance of the PAL and suggest that the Berktay's far-field solution remains to be a good model equation. (C) 2016 Elsevier Ltd. All rights reserved.

DOI： 10.1016/j.apacoust.2016.05.013

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：ACOUSTICAL SOC AMER AMER INST PHYSICS  

The parametric array loudspeaker (PAL) is an application of the parametric acoustic array in air, which can be applied to transmit a narrow audio beam from an ultrasonic emitter. However, nonlinear distortion is very perceptible in the audio beam. Modulation methods to reduce the nonlinear distortion are available for on-axis far-field applications. For other applications, preprocessing techniques are wanting. In order to develop a preprocessing technique with general applicability to a wide range of operating conditions, the Volterra filter is investigated as a nonlinear model of the PAL in this paper. Limitations of the standard audio-to-audio Volterra filter are elaborated. An improved ultrasound-to-ultrasound Volterra filter is proposed and empirically demonstrated to be a more generic Volterra model of the PAL. (C) 2016 Acoustical Society of America.

DOI： 10.1121/1.4966962

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We analyze the behaviors of active noise control using a statistical-mechanical method. The principal assumption used in the analysis is that the impulse responses of the primary path and adaptive filter are sufficiently long. In particular, in this paper we analyze the adaptation rate of the mean square error (MSE) using two measures. The first measure is the MSE initial decreasing rate. The second measure is an adaptation constant. This is defined by the negative of the maximum eigenvalue of the coefficient matrix of differential equations that describe the dynamical behaviors of the macroscopic variables. Introducing these two measures, we theoretically show that the optimal step size depends on whether we focus on the rate of decrease in the MSE at the initial stage or the MSE after sufficient adaptation time.

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：ELSEVIER SCI LTD  

The parametric loudspeaker utilizes an ultrasonic transducer array to transmit a directional sound beam in air based on the parametric array effect. In recent studies, phased array techniques have been applied to achieve controllable directivity patterns or to change the direction of the sound beam. Such a parametric loudspeaker is often referred to as a steerable parametric loudspeaker. In this paper, a dual beam generation method is elaborated. It aims to transmit two sound beams from just one steerable parametric loudspeaker. The two sound beams carries the same audio content to different locations. This dual beam generation method is compatible with the configuration of existing steerable parametric loudspeakers based on phased array techniques. As an algorithm solution, the dual beam generation method readily improves the flexibility of the steerable parametric loudspeaker. (C) 2015 Elsevier Ltd. All rights reserved.

DOI： 10.1016/j.apacoust.2015.05.004

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：ACADEMIC PRESS LTD- ELSEVIER SCIENCE LTD  

In this paper, we apply a virtual sensing technique to a head-mounted active noise control (ANC) system we have already proposed. The proposed ANC system can reduce narrowband noise while improving the noise reduction ability at the desired locations. A head-mounted ANC system based on an adaptive feedback structure can reduce noise with periodicity or narrowband components. However, since quiet zones are formed only at the locations of error microphones, an adequate noise reduction cannot be achieved at the locations where error microphones cannot be placed such as near the eardrums. A solution to this problem is to apply a virtual sensing technique. A virtual sensing ANC system can achieve higher noise reduction at the desired locations by measuring the system models from physical sensors to virtual sensors, which will be used in the online operation of the virtual sensing ANC algorithm. Hence, we attempt to achieve the maximum noise reduction near the eardrums by applying the virtual sensing technique to the head mounted ANC system. However, it is impossible to place the microphone near the eardrums. Therefore, the system models from physical sensors to virtual sensors are estimated using the Head And Torso Simulator (HATS) instead of human ears. Some simulation, experimental, and subjective assessment results demonstrate that the head mounted ANC system with virtual sensing is superior to that without virtual sensing in terms of the noise reduction ability at the desired locations. (C) 2014 Elsevier Ltd. All rights reserved.

DOI： 10.1016/j.jsv.2014.11.023

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DOI： 10.5687/iscie.28.198

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose a 3rd-order nonlinear IIR filter for compensating nonlinear distortions of loudspeaker systems. Nonlinear distortions are common around the lowest resonance frequency for electrodynamic loudspeaker systems. One interesting approach to compensating nonlinear distortions is to employ a mirror filter. The mirror filter is derived from the nonlinear differential equation for loudspeaker systems. The nonlinear parameters of a loudspeaker system, which include the force factor, stiffness, and so forth, depend on the displacement of the diaphragm. The conventional filter structure, which is called the 2nd-order nonlinear IIR filter that originates the mirror filter, cannot reduce nonlinear distortions at high frequencies because it does not take into account the nonlinearity of the self-inductance of loudspeaker systems. To deal with this problem, the proposed filter takes into account the nonlinearity of the self-inductance and has a 3rd-order nonlinear IIR filter structure. Hence, this filter can reduce nonlinear distortions at high frequencies while maintaining a lower computational complexity than that of a Volterra filter-based compensator. Experimental results demonstrate that the proposed filter outperforms the conventional filter by more than 2 dB for 2nd-order nonlinear distortions at high frequencies.

DOI： 10.1587/transfun.E98.A.820

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：ACOUSTICAL SOC AMER AMER INST PHYSICS  

This paper describes a method to compute the far-field directivity of a parametric loudspeaker array (PLA), whereby the steerable parametric loudspeaker can be implemented when phased array techniques are applied. The convolution of the product directivity and the Westervelt's directivity is suggested, substituting for the past practice of using the product directivity only. Computed directivity of a PLA using the proposed convolution model achieves significant improvement in agreement to measured directivity at a negligible computational cost. (C) 2015 Acoustical Society of America.

DOI： 10.1121/1.4906163

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

DOI： 10.11316/jpsgaiyo.70.2.0_2849

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We analyze the behaviors of active noise control with a time-varying primary path using a statistical-mechanical method. The principal assumption used in the analysis is that the impulse responses of the primary path and adaptive filter are sufficiently long. We analyze a novel model in which the reference signal is not necessarily white and the primary path is time-varying while its norm is kept constant in the mean sense. We show the existence of macroscopic steady states and the optimal step size.

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

A theory that predicts the behaviors of the Filtered-X LMS algorithm was derived by using a statistical-mechanical method. In this paper, the theory is generalized to explain the system behaviors in the case of an actual primary path. In the theory, cross-correlations between the element of a primary path and that of an adaptive filter and autocorrelations of the elements of the adaptive filter are treated as macroscopic variables. Simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables are obtained under conditions in which the tapped-delay line is sufficiently long. The equations are analytically solved to obtain the correlations and finally compute the mean-square error. In order to generalize the theory to the case of an actual primary path, the correlations of the elements of the primary path are absorbed. The generalized theory quantitatively predict the behaviors in the case of an actual primary path.

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Cambridge University Press  

A sound reproduction system usually consists of several types of loudspeakers to cater to sophisticated applications. The directivity of a loudspeaker is a measure of its efficiency in sending sounds to a particular direction instead of all directions. Demand to control the directivity of a sound reproduction system is gaining momentum with many new designs of directional loudspeakers, including the acoustic dome, horn loudspeaker, loudspeaker array, and parametric array loudspeaker (PAL). The PAL is an application of the parametric acoustic array in air, which generates a sound beam from the interaction of ultrasonic beams. The PAL has several desired features, such as its narrow beamwidth over a wide frequency range, low sound attenuation over a long distance, and ability to reproduce perceptually near sound images. The PAL is also advantageous in using a smaller driving unit to transmit an equally narrow sound beam as compared to the conventional loudspeaker and broadside loudspeaker array. An overview of directivity control methods of the PAL is presented in this paper. In particular, acoustic models and signal processing techniques in controlling the directivity of the PAL are emphasized.

DOI： 10.1017/ATSIP.2014.18

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose a parameter estimation method using Volterra kernels for the nonlinear IIR filters, which are used for the linearization of closed-box loudspeaker systems. The nonlinear IIR filter, which originates from a mirror filter, employs nonlinear parameters of the loudspeaker system. Hence, it is very important to realize an appropriate estimation method for the nonlinear parameters to increase the compensation ability of nonlinear distortions. However, it is difficult to obtain exact nonlinear parameters using the conventional parameter estimation method for nonlinear IIR filter, which uses the displacement characteristic of the diaphragm. The conventional method has two problems. First, it requires the displacement characteristic of the diaphragm but it is difficult to measure such tiny displacements. Moreover, a laser displacement gauge is required as an extra measurement instrument. Second, it has a limitation in the excitation signal used to measure the displacement of the diaphragm. On the other hand, in the proposed estimation method for nonlinear IIR filter, the parameters are updated using simulated annealing (SA) according to the cost function that represents the amount of compensation and these procedures are repeated until a given iteration count. The amount of compensation is calculated through computer simulation in which Volterra kernels of a target loudspeaker system is utilized as the loudspeaker model and then the loudspeaker model is compensated by the nonlinear IIR filter with the present parameters. Hence, the proposed method requires only an ordinary microphone and can utilize any excitation signal to estimate the nonlinear parameters. Some experimental results demonstrate that the proposed method can estimate the parameters more accurately than the conventional estimation method.

DOI： 10.1587/transfun.E97.A.2189

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose a modified-error adaptive feedback active noise control (ANC) system using a linear prediction filter. The proposed ANC system is advantageous in terms of the rate of convergence, while maintaining stability, because it can reduce narrowband noise while suppressing disturbance, including wideband components. The estimation accuracy of the noise control filter in the conventional system is degraded because the disturbance corrupts the input signal to the noise control filter. A solution of this problem is to utilize a linear prediction filter. The linear prediction filter is utilized for the modified-error feedback ANC system to suppress the wideband disturbance because the linear prediction filter can separate narrowband and wideband noise. Suppressing wideband noise is important for the head-mounted ANC system we have already proposed for reducing the noise from a magnetic resonance imaging (MRI) device because the error microphones are located near the user's ears and the user's voice consequently corrupts the input signal to the noise control filter. Some simulation and experimental results obtained using a digital signal processor (DSP) demonstrate that the proposed feedback ANC system is superior to a conventional feedback ANC system in terms of the estimation accuracy and the rate of convergence of the noise control filter.

DOI： 10.1587/transfun.E97.A.2021

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

DOI： 10.1587/essfr.8.47

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We analyze the learning curves of the active noise control with a time-varying primary path using a statistical mechanical method. The cross-correlation between the element of a primary path and that of the adaptive filter and the autocorrelations of the elements of the adaptive filter are treated as macroscopic variables. We obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under the condition that the tapped-delay line is sufficiently long. We analyze the case where the primary path has the Markovian property. As a result, we show that an optimal step size exists.

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We analyze the learning curves of the FXLMS algorithm using a statistical-mechanical method when the reference signal is not necessarily white. We treat the nonwhite reference signal by introducing the correlation function of the signal to the method proposed in our previous study. Cross-correlations between the element of a primary path and that of an adaptive filter and autocorrelations of the elements of the adaptive filter are treated as macroscopic variables. We obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under the conditions in which the tapped-delay line is long. We analytically solve the equations to obtain the correlations and finally compute the mean-square error. The obtained theory quantitatively agrees with the results of computer simulations. The theory also gives the upper limit of the step size in the FXLMS algorithm.

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We analyze the learning curves of the FXLMS algorithm using a statistical-mechanical method. Cross-correlations between the element of a primary path and that of an adaptive filter and autocorrelations of the elements of the adaptive filter are treated as macroscopic variables. We obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under the conditions in which the tapped-delay line is long. We analytically solve the equations to obtain the correlations and finally compute the mean-square error. Introducing the correlation function of the input signal, the theory can treat not only the white but also the nonwhite signal. The obtained theory quantitatively agrees with the results of computer simulations.

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose an automatic parameter adjustment method for audio equalizers using an interactive genetic algorithm (IGA). It is very difficult for ordinary users who are not familiar with audio devices to appropriately adjust the parameters of audio equalizers. We therefore propose a system that can automatically adjust the parameters of audio equalizers on the basis of user's evaluation of the reproduced sound. The proposed system utilizes an IGA to adjust the gains and Q values of the peaking filters included in audio equalizers. Listening test results demonstrate that the proposed system can appropriately adjust the parameters on the basis of the user's evaluation.

DOI： 10.1587/transfun.E95.A.2036

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Authorship：Corresponding author   Language：Japanese   Publishing type：Research paper (scientific journal)   Publisher：The Acoustical Society of Japan (ASJ)  

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Authorship：Lead author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：Cambridge University Press  

The problem of acoustic noise is becoming increasingly serious with the growing use of industrial and medical equipment, appliances, and consumer electronics. Active noise control (ANC), based on the principle of superposition, was developed in the early 20th century to help reduce noise.However, ANC is still not widely used owing to the effectiveness of control algorithms, and to the physical and economical constraints of practical applications. In this paper, we briefly introduce some fundamental ANC algorithms and theoretical analyses, and focus on recent advances on signal processing algorithms, implementation techniques, challenges for innovative applications, and open issues for further research and development of ANC systems. © The Authors, 2012.

DOI： 10.1017/ATSIP.2012.4

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

We theoretically obtain the learning curves of the FXLMS algorithm on the basis of statistical mechanical analysis. Cosines of angles between the coefficient vectors of an adaptive filter, its shifted filters, and an unknown system are treated as macroscopic variables. Assuming that the tapped-delay line is sufficiently long and exactly calculating the correlations between the past tap input vectors and the coefficient vector of the adaptive filter, we obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables in a deterministic form. We analytically solve the equations and show that the obtained theory quantitatively agrees with computer simulations. In the analysis, neither the independence assumption, the small step-size condition, nor the few-taps assumption is used.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：INST ENGINEERING TECHNOLOGY-IET  

The learning curves of the filtered-X least-mean-square (LMS) algorithm are theoretically obtained using a statistical-mechanics approach. The direction cosines among the vectors of an adaptive filter, its shifted filters, and an unknown system are treated as macroscopic variables. Assuming that the tapped-delay line is sufficiently long, simultaneous differential equations are obtained that describe the dynamical behaviours of the macroscopic variables in a deterministic form. The equations are solved analytically and show that the obtained theory quantitatively agrees with computer simulations. In the analysis, neither the independence assumption nor the few-taps assumption is used.

DOI： 10.1049/el.2011.1691

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

We propose an active noise control (ANC) system for reducing periodic noise generated in a high magnetic field such as noise generated from magnetic resonance imaging (MRI) devices (MR noise). The proposed ANC system utilizes optical microphones and piezoelectric loudspeakers, because specific acoustic equipment is required to overcome the high-field problem, and consists of a head-mounted structure to control noise near the user's ears and to compensate for the low output of the piezoelectric loudspeaker. Moreover, internal model control (IMC)-based feedback ANC is employed because the MR noise includes some periodic components and is predictable. Our experimental results demonstrate that the proposed ANC system (head-mounted structure) can significantly reduce MR noise by approximately 30 dB in a high field in an actual MRI room even if the imaging mode changes frequently.

DOI： 10.1587/transfun.E94.A.1479

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Authorship：Last author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this letter, the compensation ability of nonlinear distortions for loudspeaker systems is demonstrated using dynamic distortion measurement. Two linearization methods using a Volterra filter and a Mirror filter are compared. The conventional evaluation utilizes swept multi-sinusoidal waves. However, it is unsatisfactory because wideband signals such as those of music and voices are usually applied to loudspeaker systems. Hence, the authours use dynamic distortion measurement employing a white noise. Experimental results show that the two linearization methods can effectively reduce nonlinear distortions for wideband signals.

DOI： 10.1587/transfun.E94.A.813

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Authorship：Lead author   Language：English   Publishing type：Research paper (scientific journal)  

A novel linearization method based on MINT (Multiple-input/output INverse-filtering Theorem, was proposed using multiple loudspeaker systems. The linear and nonlinear characteristics of two loudspeakers were identified in order to conduct the compensation experiments. Since the nonlinear distortions arise around the lowest resonance frequency, the linear and nonlinear distortion compensation ability was evaluated in the frequency range from 200 to 3,000 Hz. The proposed method has low computational complexity and short delay. The number of multiplications of linear inverse filtering in the proposed method is 1,024. On the other hand, the number of multiplications of linear inverse filtering in the conventional method is 4,096. Hence, the proposed method also has an advantage with regard to system realization.

DOI： 10.1250/ast.32.220

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose a method that Uses Simulated Annealing (SA) to estimate the linear and nonlinear parameters of a closed-box loudspeaker system for implementing effective Mirror filters. The nonlinear parameters determined by W. Klippel's method are sometimes inaccurate and imaginary. In contrast, the proposed method can estimate the parameters with satisfactory accuracy due to its use of SA: the resulting impedance and displacement characteristics match those of an actual equivalent loudspeaker. A Mirror filter designed around these parameters can well compensate the nonlinear distortions of the loudspeaker system. Experiments demonstrate that the method can reduce the levels of nonlinear distortion by 5 dB to 20 dB compared to the before compensation condition.

DOI： 10.1093/ietfec/e91-a.10.3006

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In the acoustic echo canceller (AEC), the step-size parameter of the adaptive filter must be varied according to the situation if double talk occurs and/or the echo path changes. We propose an AEC that uses a sub-adaptive filter. The proposed AEC can control the step-size parameter according to the situation. Moreover, it offers superior convergence compared to the conventional AEC even when the double talk and the echo path change occur simultaneously. Simulations demonstrate that the proposed AEC can achieve higher ERLE and faster convergence than the conventional AEC. The computational complexity of the proposed AEC can be reduced by reducing the number of taps of the sub-adaptive filter.

DOI： 10.1093/ietfec/e91-a.4.1155

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In this paper, we propose a novel sound field reproduction system that uses the simultaneous perturbation (SP) method as well as two fast convergence techniques. Sound field reproduction systems that reproduce any desired signal at listener's ear generally use fixed preprocessing filters that are determined by the transfer functions from loudspeakers to control points in advance. However, control point movement results in severe localization errors. Our solution is a sound field reproduction system, based on the SP method, which uses only an error signal to update the filter coefficients. The SP method can track all control point movements but suffers from slow convergence. Hence, we also propose two methods that offer improved convergence speeds. One is a delay control method that compensates the delay caused by back-and-forth control point movements. The other is a compensation method that offsets the localization error caused by head rotation. Simulations demonstrate that the proposed methods can well track control point movements while offering reasonable convergence speeds.

DOI： 10.1093/ietfec/e91-a.3.801

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Authorship：Corresponding author   Language：Japanese   Publishing type：Research paper (scientific journal)   Publisher：The Institute of Electronics, Information and Communication Engineers  

Grant-in-Aid for Encouragement of Young Scientists

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)  

Grant-in-Aid for Encouragement of Young Scientists

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In this paper, we propose a low complexity realization method for compensating for nonlinear distortion. Generally, nonlinear distortion is compensated for by a linearization system using a Volterra kernel. However, this method has a problem of requiring a huge computational complexity for the convolution needed between an input signal and the 2nd-order Volterra kernel. The Simplified Volterra Filter (SVF), which removes the lines along the main diagonal of the 2nd-order Volterra kernel, has been previously proposed as a way to reduce the computational complexity while maintaining the compensation performance for the nonlinear distortion. However, this method cannot greatly reduce the computational complexity. Hence, we propose a subband linearization system which consists of a subband parallel cascade realization method for the 2nd-order Volterra kernel and subband linear inverse filter. Experimental results show that this proposed linearization system can produce the same compensation ability as the conventional method while reducing the computational complexity.

DOI： 10.1093/ietfec/e90-a.8.1616

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose the introduction of a frequency domain variable perturbation control and a leaky algorithm to the frequency domain time difference simultaneous perturbation (FDTDSP) method in order to improve the cancellation performance and the stability of the active noise control (ANC) system using the perturbation method. Since the ANC system using the perturbation method does not need the secondary path model, it has an advantage of being able to track the secondary path changes. However, the conventional perturbation method has the problem that the cancellation performance deteriorates over the entire frequency band when the frequency response of the secondary path has dips because the magnitude of the perturbation is controlled in the time domain. Moreover, the stability of this method also deteriorates in consequence of the dips. On the other hand, the proposed method can improve the cancellation performance by providing the appropriate magnitude of the perturbation over the entire frequency band and stabilizing the system operation. The effectiveness of the proposed method is demonstrated through simulation and experimental results.

DOI： 10.1093/ietfec/e90-a.8.1555

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：SCRIPTA TECHNICA-JOHN WILEY & SONS  

In this paper, a method using a subadaptive filter playing the supplementary role of acoustic echo cancellation is discussed. Since the updating state of the adaptive filter can be calculated approximately as a characteristic of the subadaptive filter, the step size parameter can be optimally controlled depending on the estimation accuracy. Also, for prevention of echo path variation, we discuss a variable detector in which the variation of the filter coefficients at the time of variation is considered. The proposed method is found to be robust to channel variations and can improve the ERLE by 10 to 15 dB in comparison with the conventional method. (C) 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(2): 9-21, 2007;

DOI： 10.1002/ecjc.20231

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Authorship：Corresponding author   Language：Japanese   Publishing type：Research paper (scientific journal)   Publisher：The Institute of Electronics, Information and Communication Engineers  

CiNii Books

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

CiNii Books

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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The subband adaptive Volterra filter (SBAVF), which can assign the filter length of each subband arbitrarily, is proposed. If the frequency characteristics of a system to be identified are shifted to a certain bandwidth, it is usually not possible to carry out efficient identification by taking account of the shift of the bandwidth in a (full band) adaptive Volterra filter. On the other hand, in the proposed SBAVF, efficient system identification is possible by assigning a longer filter length to the dominant frequency range of the system of interest. When the proposed method is applied to a speaker system, it is shown through simulation results that the error attenuation can be improved from 1.5 dB to 7.5 dB in comparison with the conventional method. © 2004 Wiley Periodicals, Inc.

DOI： 10.1002/ecjc.10135

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In this paper, an automatic method for the design of compact acoustic systems, using the genetic algorithm, is proposed for improvement of degraded sensitivity caused by leakage. In the present method, an analysis based on the acoustic equivalent circuit is used. The equivalent circuit and the acoustic parameters of the circuit are designed in such a way that the sound pressure frequency characteristics are contained with the range based on the GSM (ETSI EN 300 903) specifications and that the sensitivity deterioration is improved. The PfGA (Parameter-free Genetic Algorithm) is used in the design of the circuit configuration, and the genetic algorithm of Michalewicz is used for the design of acoustic parameters. As a result, an equivalent circuit is obtained that realizes characteristics not significantly affected by leakage. The present method can be applied to various compact acoustic systems such as cellular phones and provides useful information for actual design.

DOI： 10.1002/ecjc.10149

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Language：Japanese   Publisher：The Acoustical Society of Japan (ASJ)  

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An active noise control (ANC) system is proposed that uses a new frequency domain simultaneous perturbation (FDSP) method automatically controlling the magnitude of perturbation. The proposed method automatically controls the magnitude of perturbation so that the convergence speed can be improved substantially over the conventional case where the magnitude of perturbation is fixed. Specifically, when the error signal is large, the magnitude of the perturbation is made large, while the latter is reduced when the error signal is small. As a result, the present system resolves the problem of instability caused by the block process and also the step size can be set to a larger value. The present control is effective when the magnitude of perturbation needs to be set small. In particular, if the magnitude of perturbation is 0.08, the convergence speed is improved by more than 50 times. Also, the present control is effective for suppression of the noise generated due to addition of perturbation.

DOI： 10.1002/ecjc.10112

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科研費奨励研究

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we present three kinds of effectiveness of the FDSNBLMS algorithm with block length control. The first effectiveness is that the proposed algorithm can improve the convergence speed and realize the low computational complexity. Concretely, the proposed algorithm can reduce the convergence time by half compared with the conventional algorithm in order to obtain the same estimation accuracy. Moreover, the execution of 2-D FFT can be reduced in proportion to sample time. The second effectiveness is that the proposed algorithm can make the convergence fast and stable compared with the conventional algorithm for music signal. Concretely, the estimation accuracy of the proposed algorithm is two times higher than that of the conventional algorithm in the same sample time. The third effectiveness is the robustness. Concretely, the proposed algorithm can cope with noise and impulse response changes caused in common environments.

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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.

DOI： 10.1002/ecjc.10000

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a design method of compact acoustic systems for preventing the sensitivity decline due to leaks by the Genetic Algorithm. This method can design acoustic equivalent circuits and the corresponding acoustic parameter values whose Sound Pressure Level is within the range based on the GSM(ETSI EN 300 903) and prevent the sensitivity decline. The proposed method uses the PfGA (Parameter-free Genetic Algorithm) for the design of equivalent circuits and the Michalewicz's Genetic Program for the design of acoustic parameters. We can obtain some equivalent circuits whose frequency response is hardly affected by leak. This method can be used for compact acoustic systems such as cellular phones. Furthermore, you can obtain useful information from the design results.

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In the problem of restoration of images degraded by impulsive noise, a method has been proposed to eliminate impulsive noise by using the noise position image formed with noise detectors. However, such a method estimates the signal values of the pixels to be processed by using only the nondegraded pixels within the filter window. Hence, if the majority of the window is occupied by impulsive noise, the signal information needed for estimation becomes insufficient and the estimation accuracy is significantly reduced. In addition, in the previous method, average value filters and median filters are used for estimation of the signal value of the processing points. No processing has been included that takes account of the directionality (shape) of image signals such as edge components and detail components. In the present paper, a new noise elimination filter is proposed that attempts to improve restoration accuracy by addressing these two problems. In the proposed filter, first the degraded pixels within the filter window are precompensated for the problem of insufficient original signal pixels in the filter window, and then the signal values at the processing points are estimated by the filter once sufficient signal information is obtained. Preservation of the directionality of the image signals is improved by extending the directional difference filter. Its effectiveness is verified by real image processing results. © 2003 Wiley Periodicals, Inc.

DOI： 10.1002/ecjc.1149

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a subband adaptive Volterra filter, whose filter taps can be appropriately assigned at each subband. The proposed subband adaptive Volterra filter can effectively identify nonlinear systems with nonlinear distortions concentrated in a frequency band. In this case, the conventional adaptive Volterra filter is ineffective because such a system may require long filter taps at a frequency band and short filter taps at another frequency band. Simulation results demonstrate the effectiveness of the proposed subband adaptive Volterra filter. In the simulation, we identify a loudspeaker system by using the proposed and conventional adaptive Volterra filters with the same computational complexity. As a result, the Reduction of the proposed adaptive Volterra filters is from 1.5dB to 7.5dB higher than that of the conventional one.

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this letter, we propose a frequency domain active noise control system using the time difference simultaneous perturbation method. This method is an algorithm based on the simultaneous perturbation method which updates the coefficients of the noise control filter only by use of the error signal. The time difference simultaneous perturbation method updates the filter coefficients by using one kind of error signal, while the simultaneous perturbation method updates the filter coefficients by using two kinds of error signal. In the ANC systems, the time difference simultaneous perturbation method is superior because ANC systems cannot obtain two error signals at the same time. When this method is applied to ANC systems, the convergence speed can be increased to a maximum of twice that of the conventional method.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose an active noise control (ANC) system using a novel frequency domain simultaneous perturbation method with automatic perturbation control. The proposed method can converge faster than the conventional method that uses the fixed perturbation. Concretely, the present perturbation value is set in proportional to the present error signal. As a result, the proposed system can improve the instability due to block processing and set the step-size large. The proposed system has the effectiveness when the perturbation value must be made small, especially, the proposed method can converge 50 times as fast as the conventional method in case where the perturbation value is 0.08. Moreover, the proposed method can suppress the noise due to the perturbation.

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：SCRIPTA TECHNICA-JOHN WILEY & SONS  

Since a speaker system has nonlinearity, it is necessary to place a nonlinear inverse system in front of the speaker system so that the nonlinear distortion is corrected. However, in realization with DSP, the amount of computation in the Volterra filter used in the nonlinear inverse system is enormous. There exists a region containing aliasing components in the output signal of the Volterra filter. There is a way to band-limit the input signal in order to avoid aliasing. However, since the sampling frequency is set to twice the Nyquist frequency, an unnecessary band is processed and the efficiency is poor. Hence, in this paper, a configuration is proposed that does not cause aliasing even if the sampling frequency is chosen as the Nyquist frequency in order to reduce the amount of computation in the Volterra filter. Next, an identification method for the speaker system in order to efficiently obtain the Volterra filter used in the proposed configuration is presented. The proposed configuration is applied to the nonlinear inverse system. Finally, the effectiveness of the present method is demonstrated by simulation. (C) 2002 Wiley Periodicals, Inc.

DOI： 10.1002/ecjc.10031

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：Acoustical Society of Japan  

In this paper, we present the effectiveness of a software tool for eliminating nonlinear distortions of loudspeaker systems through some experiments. The nonlinear distortions affect the sound quality of loudspeaker systems. We have presented some identification methods of loudspeaker systems and some compensation methods of nonlinear distortions. However, the software tool for compensating the distortions has not been realized yet. We therefore develop the software tool. Experimental results show that the 2nd- and 3nd-order nonlinear distortions of a loudspeaker system can be reduced in the range of 10 [dB] to 20 [dB] by the developed tool when noise level is below 70 [dBA].

DOI： 10.1250/ast.24.186

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this letter, we propose an improvement of the overall online modeling method in active noise control (ANC) systems. The proposed method utilizes the summational normalized LMS algorithm with a block length control for modeling a secondary path. The improved overall online method can consequently determine the step-size parameter more easily and converge faster than the conventional one. Moreover, the tracking property for variations of physical systems is improved. Simulation results demonstrate the fast convergence and the robustness for the variations of physical systems.

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Authorship：Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：SCRIPTA TECHNICA-JOHN WILEY & SONS  

In this paper, a new median filter is proposed that has robustness without relying on the noise generation probability. The proposed filter has a configuration with an impulse detector in the prior stage of the filter. Since this detector has a superior signal estimation capability of the directional difference filter, impulse noise can be detected effectively even if generated at a high generating ratio and as a burst. In the median filter in the later stage, inclusion of the output values of the directional filter and successive updating of the signal values (successive process) are introduced. Hence, while the noise elimination capability of the conventional median filter is retained, the signal preservation at the details and edges can be improved. Therefore, the noise elimination filter proposed in this paper is abundant in robustness against variations of the gray scale, the generation probability, and the impulse adding situation. Its restoration results for images with superposed impulse noise at a high generation probability confirm the improvement of performance. In this paper, the effectiveness of the proposed impulse detector is first demonstrated. It is proven through processing examples that a better image processing result can be obtained from its combination with a newly configured median filter. At the same time, the robustness of the proposed procedure under various conditions is demonstrated. The effectiveness of the proposed method in the final restored image is verified. (C) 2001 Scripta Technica, Electron Comm Jpn Pt 3, 85(3): 22-32, 2002.

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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.

DOI： 10.1002/ecjc.1124

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Language：English   Publishing type：Research paper (scientific journal)  

In this paper, we propose a directional difference filter that possesses superior noise removal for images corrupted by additive impulse noise and a high level of original signal preservation. By estimating the pixel value of the processing point from directional shape information about the edges and successively updating the estimate, the proposed filter can prevent residual noise even when impulse noise generated at a high probability was added. In addition, this process can perform original signal preservation robust to the detection errors of the noise detector and is extremely useful as a postprocess for a noise detector that is used to remove noise generated at a high probability. In this paper, we demonstrate the robustness of the proposed method to the detection errors generated when using an actual noise detector and use processing examples to verify the superior image processing results over conventional noise removal filters. © 2001 Scripta Technica, Electron. Comm. Jpn. Pt.

DOI： 10.1002/ecjc.1073

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Authorship：Lead author,　Corresponding author   Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we propose a frequency domain active noise control (ANC) system without a secondary path model. The proposed system is based on the frequency domain simultaneous perturbation (FDSP) method we have proposed, In this system, the coefficients of the adaptive filter are updated only by error signals. The conventional ANC system using the filtered-x algorithm becomes unstable due to the error between the secondary path, from secondary source to error sensor, and its model. In contrast, the proposed ANC system has the advantage not to use the model. In this paper, we show the principle of the proposed ANC system, and examine its efficiency through computer simulations.

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We propose a music retrieval system with words expressing impression of the tune (sensitivity words) in this paper. The conventional music retrieval systems with sensitivity words have the following problems: (1) they cannot retrieve music with delicate nuance; (2) they cannot retrieve music with more than one sensitivity word; (3) the orthogonal of axes forming the retrieval space is not guaranteed; (4) they asume the relation between tune impression and sensitivity words or tune charactseristics to be linear; (5) only objective or subjective evaluation goes in the evaluation of the system. In this paper, we therefore propose a novel music retrieval system in order to solve these problems. First, the retrieval space, which expresses the psychological correlation of tune impression, is constructed by using the semantic differential method and the factor analysis. Moreover, a position of the tune registered in the music database on the retrieval space is obtained by the genetic algorithm, the multiple regression analysis, and the neural network. Next, a position corresponding to the impression input by user on the retrieval space is predicted by the multiple regression analysis or the neural network, then the tune positioning near the predicted position is output as the retrieval result. Finally, the objective evaluation demonstrates that the correct rates of the proposed system are 100% and the subjective evaluation demonstrates that the proposed system satisfied the 90% subjects.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose the method identifying the 3rd-order Volterra kernels by multi-sinusoidal waves. The level of the 3rd-order nonlinear distortion of loudspeaker systems is nearly equal to that of the 2nd-order one. The 3rd-order nonlinear distortion affects the sound quality of loudspeaker sys-tems. Therefore, the 3rd-order nonlinear distortion should be eliminated. Identifying the 3rd-order nonlinear distortion with the 3rd-order Volterra filter, however, is essential to eliminate it. In this pa-per, the features of the 3rd-order Volterra kernels are explained and the identification method using those features is proposed. We also identify the Volterra kernels of an actual loudspeaker system by the proposed method and examine the identification accuracy. Moreover, we eliminate the 3rd-order nonlinear distortion by using the identified Volterra kernels and demonstrate the effectiveness of the proposed method.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a novel progressive switching (PS) based filter. The proposed filter compensates corrupted pixels except the processing pixel by using all pixels (including the compensated pixels) in the same window. In contrast, the conventional PS filters estimate the processing pixel by only uncorrupted pixels (original pixels). Hence, the proposed filter has higher estimation accuracy than the conventional one in case of being lacking in available uncorrupted pixels. This case occures when most pixels are corrupted by impluse noise in the processing window. The proposed filter also incorporates shapes (directions) of image signals, such as edges and details, into the filtering operation. Hence, the proposed filter can preserve original pixels better than the conventional one. The filtering operation of the directional difference filter we have already proposed in applied to the proposed filter in order to realize the above operation. We examine the efficiency of the proposed filter through a lot of examples. Experimental results demonstrate that the proposed filter can improve the noise removal ability and original signal preservation.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Authorship：Lead author,　Corresponding author   Language：Japanese   Publishing type：Research paper (scientific journal)   Publisher：IEEE  

In this paper, we propose a frequency domain active noise control system using optimal step-size parameters at each frequency. The proposed ANC system call converge faster than the conventional ANC system using the Filtered-x LMS algorithm with the optimal stepsize parameter. Moreover, the proposed system can converge by setting the step-size parameters at unstable frequencies to 0 in the case where the phase error of the secondary path model does not satisfy the stable condition, whereas the conventional ANC system cannot converge in this case. In this paper, the theoretical equation of the optimal step-size parameters is derived by using available information during system operation. Next, we present the structure of the ANC system using the optimal step-size parameters obtained from the theoretical equation. Moreover, a control technique determining unstable frequencies is introduced. Finally, simulation results demonstrate the efficiency of the proposed ANC system.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a directional difference filter, which has the superior noise removal ability for images degraded by impulsive noise and the original signal preservation. The proposed filter estimates the value of pixel at a processing point from directional form information of edge in the local domain and restores that value at the point consecutively so as to be able to remove impulsive noise with high probability. The proposed method can preserve the original signal robustly for the detection error of noise detector. This filter works effectively as the postprocessor of noise detector used for removing a high generating probability noise. In this paper, we show that the proposed method has the robustness for the detection error of actual noise detectors and the superiority to a conventional noise reduction filter on image processing from experiments.

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In this paper, we propose the component of a Neural Filter that copes with power variations of input signals. Proposed Neural Filter consists of an Adaptive Filter and two types of Neural Filters. The Adaptive Filter identifies linear properties, one of Neural Filters identifies nonlinear wave properties, and another Neural Filter identifies nonlinear amplitude properties of the target system. Therefore proposed Neural Filter is suitable for identifying the nonlinear system that has power variations of input signals such as a loudspeaker system. In this paper, it is showed that proposed Neural Filter copes with power variations of input signals and has the higher accuracy than even scale Adaptive Volterra Filter for identifying third order nonlinear system. These results are showed on computer simulations.

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Modeling loudspeaker systems by Volterra series expansion is essential to eliminate the nonlinear distortion. We have proposed a method measuring the Volterra kernel of loudspeaker systems by multi sinusoidal wave. However, this method has problems not to consider the phase property of nonlinear element, third-order distortion, and over, the gap of amplitude and phase by using anti-aliasing filter. Therefore, we propose the new method measuring the second-order Volterra kernel by multi sinusoidal wave. In this method, the phase property of nonlinear element, third-order distortion, and the gap by using anti-aliasing filter are considered. And, we eliminate the nonlinear distortion of loudspeaker systems with the Volterra kernel measured by the new method on offline, and show the effectiveness of the new method.

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Publishing type：Research paper (scientific journal)  

Recently, the methods of using the digital filter (Volterra filter) based on the Volterra series expansion for the identification of a nonlinear system and the elimination of a nonlinear distortion have been proposed. The Volterra filter is obtained by extending conventional linear digital filter into nonlinear, and the structure and the characteristic include the linear digital filter. Therefore, the methods that have been developed in the linear digital filter can be easily introduced. In this paper, I introduce adaptive Volterra filters that modify the filter coefficients automatically. Especially, its update algorithms and problems are introduced, and its future is mentioned.

DOI： 10.1002/1520-6440(200012)83:12<51::AID-ECJC6>3.0.CO;2-K

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Scripta Technica Inc  

The distortion of the speaker system, as the terminal end of an audio playback system, can be divided into linear distortion and nonlinear distortion. The authors have carried out research on elimination of linear distortion by using digital filters. In such a process, when the equalization effect at low frequencies is increased, the amplitude of the speaker vibrating plate increases so that the nonlinear distortion increases. Therefore, in order to construct a high-fidelity audio playback system without distortion, it is necessary to eliminate nonlinear distortion in addition to linear distortion. To this end, the Volterra series that describes the nonlinear input-output relationship is used for identification of the audio playback system. A procedure is proposed to design an inverse system. In addition, a simulation is carried out. As a result, it is possible to reduce the nonlinear distortion of a speaker system by as much as 100 dB by the proposed method, confirming the usefulness of the method.

DOI： 10.1002/(SICI)1520-6440(200002)83:2<110::AID-ECJC10>3.0.CO;2-X

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Scripta Technica Inc  

In this paper, we propose an online design method of a nonlinear inverse system which eliminates the linear and nonlinear distortions of the nonlinear system at the same time. We discuss the online design method by using the adaptive Volterra filter. Although we have already proposed offline design methods, these had the problem that the elimination level of the nonlinear distortion degrades when the characteristics of the target nonlinear system vary. In this paper, we first show the effects of compensation of distortions due to the modeling error of each filter which composes a nonlinear inverse system. Next, the proposed online design method is explained by dividing it into three steps. In the first step, the online identification method of the first-order Volterra kernel is discussed. In the second step, the online design method of the linear inverse filter is discussed. In the third step, the online identification method of the second-order Volterra kernel is discussed. Finally, the effectiveness of the proposed method is shown by computer simulations.

DOI： 10.1002/(SICI)1520-6440(200009)83:9<46::AID-ECJC5>3.0.CO;2-L

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：Scripta Technica Inc  

In this paper, we propose an active noise control (ANC) system without secondary path model (C 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.

DOI： 10.1002/(SICI)1520-6440(200010)83:10<47::AID-ECJC6>3.0.CO;2-T

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we discuss a formulation of the convergence property for the second-order Volterra filter using NLMS algorithm. Many studies of the convergence property for linear adaptive filter have been tried up to now. However, the convergence property of the adaptive Volterra filter has been never analyzed. Therefore, we clarify the convergence property for the second-order Volterra filter theoretically by formulating the convergence property of E-NLMS algorithm for the adaptive Volterra filter. We use the first order recursive filter expression, which is effective to analyze various properties of NLMS algorithm for linear adaptive filter, to formulate the convergence property of E-NLMS algorithm, and moreover, use the characteristic of the second order Volterra filter which is divided into the diagonal elements (D elements) and the other (ED elements). We show the theoretical formula derived in this paper to be effective by computer simulation.

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In this paper, we derive the convergence condition of the simultaneous perturbation optimization method applied to adaptive filter. The adaptive filter using the simultaneous perturbation optimization method has the characteristic updating the filter coefficients of adaptive filter by adding the perturbations to the filter coefficients. Hence, the simultaneous perturbation optimization method is expected to apply to active noise control (ANC) systems and pre-inverse systems.However, the convergence condition of the simultaneous perturbation optimization method applied to adaptive filter has not been studied up to now. In this paper, we derive theoretically the upper limit of step size parameter needed in order to converge the adaptive filter in the simultaneous perturbation optimization method. In addition, simulation results demonstrate the effectiveness of the derived theoretical formulae.

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Language：Japanese   Publisher：Information Processing Society of Japan (IPSJ)  

In this paper, we propose a method which can extract tune's characteristic and impression automatically and classify into the arbitrary groups only from score information given to computer using the nonlinear distinction analysis. The try to imitate a player's music activity by computer is called music information processing, and related researches have been studied actively in recent years. In these researches, the following methods are reported as a solution of the proposition how the player interprets a score, and gives the performance; (1) The rule generation method by production system considering the information described in the score and musical theory. (2) The method using neural network. (3) The method using multiple regression analysis. However, in spite of the fact that the skilled player changes his or her playing style according to the characteristic and impression of the tune, the system considering the fact has not existed yet. Therefore in this paper, in order to settle up the problem, the psychological listening experiment based on the standpoint of actual listeners is done using the SD method and the factor analysis at first. And applying the nonlinear distinction analysis by the neural network to the result, the computer automatically distinguishes what impression the tune has, and distributes the tune to a suitable group according to the impression. Furthermore, in an unknown tune, this system succeeded in the distinction of the impression of the tune by high accuracy of 88%.

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Other Link： http://id.nii.ac.jp/1001/00012753/

Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose an active noise control (ANC) system without secondary path model. The proposed system is based on the simultaneous perturbation optimization method with block process. Consequently, the coefficients of an 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 secondary source to error sensor and its model. On the other hand, the proposed ANC system has an 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.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose an online design method of a nonlinear inverse system that eliminates the linear and the nonlinear distortions of a nonlinear system at the same time. We discuss the online design method by using adaptive Volterra filter. Although we have already proposed the off-line design methods, these had a problem that the elimination level of nonlinear distortion degrades when the characteristic of a target nonlinear system varies. First of all in this paper, we show effects of elimination of distortions due to modeling error of each filter that composes a nonlinear inverse system. Next, the proposed online design method is explained by dividing into three steps. At the first step, the online identification method of first order Volterra kernel is discussed. At the second step, the online design method of linear inverse filter is discussed. At the third step, the online identification method of second order Volterra kernel is discussed. Finally, simulation results demonstrate the effectiveness of the proposed method.

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Publishing type：Research paper (scientific journal)  

The active noise control system using the strictly positive real property of the error model in the adaptive control theory has been proposed recently. However, this method is of no practical use because it has the assumption that the error path between the secondary source and the error sensor are known. In this paper, we present the convergence property of this method without the assumption above mentioned.We make the convergence property clear by using the modified block diagram and the T'th order IIR filter expression. It is made clear that the disturbance preventing the estimation of filter coefficients varies with its probability. Consequently, we make it clear that the convergence property of this method has danger of divergence.

DOI： 10.1002/(SICI)1520-6440(199909)82:9<18::AID-ECJC3>3.0.CO;2-W

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：John Wiley and Sons Inc.  

When an ANC system is operated with the Filtered-x LMS algorithm, the step size parameter constant needs to be determined. If this constant is too small, the convergence of the filter coefficient becomes slow. If, on the other hand, it is increased too much to speed up the convergence, the system becomes unstable. It is necessary to find the upper limit for stable operation so that system stability is satisfied and convergence is fast. Such a value has hitherto been unknown. Therefore, the step size parameter has been determined by trial and error. For the Filtered-x LMS algorithm, we need a filter Ĉ that identifies the error path C between the secondary speaker in the ANC system and the error detection microphone. Naturally, modeling error is contained in this filter. In this paper, a theoretical equation for the upper limit of the step size parameter for stable operation is derived from the phase: error of the allowable C and Ĉ. We also attempt to represent this theoretical equation in terms of information obtainable at the time of system operation. The convergence characteristics in the case using the step size parameter value obtained from the theoretical equation and in the case using other values are compared, and the effectiveness of the proposed theoretical equation is demonstrated. © 1999 Scripta Technica.

DOI： 10.1002/(SICI)1520-6440(199911)82:11<65::AID-ECJC8>3.0.CO;2-Q

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：John Wiley and Sons Inc.  

The distortion of a speaker system at the end of an audio playback system can be divided into linear distortion and nonlinear distortion. The authors have carried out research on elimination of linear distortion by using digital filters. In this process, when the equalization effect at low frequencies is increased, the amplitude of the speaker vibrating plate increases so that the nonlinear distortion increases. Therefore, in order to construct a high fidelity audio playback system without distortion, it is necessary to eliminate nonlinear distortion in addition to linear distortion. To this end, a Volterra series that describes the nonlinear input-output relationship is used to identify the audio playback system. A procedure for the design of an inverse system is proposed. In addition, a simulation is carried out. It was found possible to reduce the nonlinear distortion of a speaker system by as much as 100 dB by using the proposed method. © 1998 Scripta Technica.

DOI： 10.1002/(SICI)1520-6440(199901)82:1<87::AID-ECJC10>3.0.CO;2-5

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：SCRIPTA TECHNICA-JOHN WILEY & SONS  

A conventional linear filter increases nonlinear distortion when it is applied to the elimination of linear distortion of a system. This paper proposes an off-line design method of an adaptive Volterra filter that eliminates linear and nonlinear distortions at the same time. This paper analyzes, by using the Volterra theory, the mechanism of increase of nonlinear distortion due to the elimination of linear distortion. This paper also analyzes, again using the Volterra theory, a signal component caused by the tandem connection of a nonlinear system and gives a design method for a nonlinear inverse system that eliminates nonlinear distortion. The proposed method reduces the nonlinear distortions by as much as 70 dB. The design time of the method is much shorter than that of existing conventional methods. (C) 1997 Scripta Technica, Inc.

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In this paper, we pay attention to the NLMS algorithm, which is one of algorithms for the adaptive Volterra filter (AVF), and derive the convergence and the stability conditions. As a result, we show that both of the conditions in the nonlinear adaptive filter are the same as the condition in the linear, and also that each estimation error of the first- and second-order AVF affects another AVF in the systems that compose the first- and second-order AVF.

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Language：Japanese   Publisher：The Acoustical Society of Japan (ASJ)  

The design in which acoustic equivalent circuit is used to design the telephone-handset is suitable for achieving given design philosophy. However, the designed acoustic parameters were impossible to be converted into actual sizes because theoretical formulae (relations) between the acoustic parameters in the acoustic equivalent circuit and the sizes of cavities and small holes, don't exist. Therefore, the method of modeling the above-mentioned relations by the neural network is proposed in this paper. In this method, the neural network learns the above-mentioned relations by using many pairs of acoustic parameters and actual sizes obtained by changing the sizes of cavities and small holes in telephone-handset variously. The acoustic parameters designed to achieve design philosophy given in the acoustic equivalent circuit can be converted into actual sizes according to this method. Moreover, since the scale of the neural network used to learn is small, the characteristics that learning time is short and the accuracy of the output value is high is possessed.

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In this paper, the off-line design method of a nonlinear inverse system, which eliminates linear distortion and nonlinear distortion at the same time, is proposed. The proposal method is a design method in time domain using adaptive Volterra filter. The linear inverse system with conventional linear filter has the fault that the nonlinear distortion increases though the system can eliminate the linear distortion. In this paper, the mechanism that the nonlinear distortion increases with elimination of the linear distortion is theoretically clarified by the Volterra theory. And, the design method of nonlinear inverse system, which eliminates the nonlinear distortion, is given by deriving the signal component caused by the tandem connection of the nonlinear system by the Volterra theory. The proposal method realizes that the nonlinear distortion can be decreased by 70dB or less and shortening of the design time is also possible.

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There are two types of distortion of the loudspeaker system, the linear and the nonlinear distortions. We have tried to eliminate the linear distortion by digital filters. However, the more quantity of the elimination of the linear distortion, results in the more quantity of the nonlinear distortion. Therefore in order to make a high quality audio system we need to eliminate both the linear and the nonlinear distortions at the same time. First, we have identified the loudspeaker system by the Volterra expansion, which expresses the relationship between input and output signals, and then proposed the design method of the inverse system. The result of simulation shows that a decrease of the nonlinear distortion of about 100dB can be obtained by using the design method proposed here. This shows the utility of the proposed method.

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

When we use a telephone-handset, the frequency response of the telephone-earphone becomes degraded because of the leak through the slit between the ear and the earphone. Consequently, it is very important to establish the design method of the telephone-handset which reduces the effect of leak. No one has tried to design the telephone-handset to reduce the effect. We are the only ones to have proposed an automatic design method by nonlinear optimization techniques. However, this method gives only one set of the acoustic parameters aiming at a certain specific target frequency response, and therefore lacks flexibility in the actual design problem. On the other hand, the design method proposed in this paper, which uses Monte-Carlo method, gives an infinite number of sets of acoustic parameters that realize infinite frequency responses within the target allowable region. As these infinite number of sets become directly the design ranges of acoustic parameters, the proposed method has the flexibility that any set of the acoustic parameters belonging to the design ranges guarantees the corresponding response to be within the target allowable region, and at the same time reduces the effect of leak. This flexibility is advantageous to the actual design problem.

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One design method of the acoustic system is to obtain the actual sizes after determining the acoustic parameters that realizes a target frequency response. However, the compact acoustic system, such as telephone-handset, is designed by cutting and trying at present, because the analysis of the acoustic equivalent circuit becomes very complicated. The factors making the analysis of the equivalent circuit complicated are the following two points: 1) the formula and the theory relating the acoustic parameter with the actual size are not established; 2) the number of parameters and the degree of freedom increase if the position and the number of the small holes are taken into account in the acoustic equivalent circuit. Therefore, these problems should be solved or avoided. Here we propose the design method which can obtain directly the sizes of structure from the frequency response without the analysis of the acoustic equivalent circuit. That is, this method can design directly the sizes of structure without using the equivalent circuit, because this method can model the relation between the actual size and the frequency response by the neural network. The result of evaluating the neural network constructed here suggests that this method is very effective for designing telephone-handsets.

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Language：Japanese   Publisher：The Acoustical Society of Japan (ASJ)  

When we really use a telephone handset, the listening voice leaks through the slit between the ear and the earphone, and the transmission quality becomes poor. For reducing this effect by the leak, up to now telephone designers have reformed the shape of ear-piece, or telephone users have pressed the earpiece tightly against their ears. However these methods have a limit actually. Therefore, telephone designers should consider the structure of telephone handset such as reduces the effect of the leak in advance. In this paper we proposed an automatic design method, based on the nonlinear optimization technique, of such a telephone handset as reduces the effect of the leak by considering an estimation function applicable to various quantity of leak. As a result the diaphragm needs to be soft, light and resistive, and the capacity of cavities needs to be bigger than original one. Some examples of trial production show that the design method proposed here is effective to reduce the effect of leak.

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Language：English   Publishing type：Research paper (scientific journal)  

A problem in the telephone handset is that there is no design formula that can relate the values of the parameters in the acoustic equivalent circuit (acoustic parameters) of the earphone system directly to the shape (dimensions) of the actual handset. As a result, the design of the handset has been made very difficult. to derive such a formula, the values of the acoustic parameters must be estimated accurately from the actual measurement of the handset. Previously, the authors have estimated that acoustic parameters, using the nonlinear optimizatoin technique, but a problem is still left that the number of parameters that can be simultaneously estimated is limited. This paper tries to solve this problem and proposes and estimation technique using the Monte Carlo method, which can also realize a high estimation accuracy. In the proposed method, an allowable range for the actual response is defined and the acoustic parameters are estimated by modifying the upper and the lower limits of each acoustic parameter so that the response stays within the allowable range. The proposed method has the advantages that the number of parameters to be estimated is not limited, a good agreement can be realized with the actual response, and the decision can be made without intuition. the approach also provides a useful means for the design of the whole system. Although this paper discusses the estimation for the piezoelectric telephone handset, the proposed method can be applied effectively to any type of telephone handset. Copyright © 1995 Wiley Periodicals, Inc., A Wiley Company

DOI： 10.1002/ecjc.4430780905

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：SINTEF  

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Other Link： https://ndlsearch.ndl.go.jp/books/R000000004-I028315230

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Language：Japanese   Publisher：関西大学先端科学技術推進機構  

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

DOI： 10.11316/jpsgaiyo.70.2.0_2849

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Language：English   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Feedforward active noise control (ANC) system can reduce various noise such as the broad-band noise by setting the reference microphone close to a noise source. However, the reference microphones cannot be set close to the noise sources in case where various noises propergare from the various directions or noise source moves. In these cases, the performance of ANC system deteriorates. Therefore, we propose an ANC system with noise source localization in this paper. In this system, some microphones located around the noise control point are used to estimate noise source locations, and the nearest microphone to the noise source is used as the reference microphone to reduce the noise propagating from the noise source. We estimated noise source locations actually and tried noise reduction in the proposed ANC system.

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Language：Japanese   Publisher：[東京] : [電子情報通信学会]  

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Other Link： https://ndlsearch.ndl.go.jp/books/R000000004-I025638909

Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

This paper proposes an automatic detection and tracking system for moving objects based on Kinect and PTZ (Pan-Tilt-Zoom) camera. The microphone array in Kinect is used for detecting the moving object, and the PTZ camera tracks it. In the tracking, particle filter and real adaboost are adopted. If an object is the outside of the scope of the camera, we can easily aim the scope of the camera at the object. The experimental result under the real environment shows the effectiveness of the proposed system.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Acoustic noise problems are becoming increasingly serious in the process of industrialization. In particular, an effective approach to reduce unwanted noise is needed in factory. Therefore, we have studied an active noise control (ANC) system to reduce factory noise where manufacturing equipments generate noise levels of over 90 dB. In this paper, we focus on a multi-channel ANC system using parametric array loudspeaker (PAL) as control sources. The proposed ANC system can not only reduce unwanted acoustic noise at the desired locations but also suppress the increase in sound pressure level at other locations because the PAL have a super-directivity feature. If interferences between different channels of the proposed ANC system are minimized, therefore each channel in this ANC system can be controlled independently because PAL are used as secondary sources. By this mean, the computational complexity of a multiple-channel ANC system is reduced since the cross-talk secondary path models are negligible. In this paper, we demonstrate the infuence of cross-talk in proposed Case(1,2,2,) ANC syetem. The performance of the proposed ANC system is demonstrated through experiments and simulations. It has been confirmed that PAL outperformed omnidirectional loudspeakers in terms of being free from cross-talk in the Case(1,2,2) ANC system.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine the effect of spatial relationship of acoustic holes to acoustic characteristics by using Finite Element Method(FEM). We quantitatively clarify the effect on acoustic characteristics from spatial relationship between acoustic holes using the acoustic parameter estimation method we developed. A new design guideline for compact acoustic reproduction systems are provided through examining the relationship between simulation of electric impedance characteristics and the corresponding experimental results.

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

We have studied an ANC (Active Noise Control) system for reducing MR noise, which is generated from MRI device and realizing clear verbal communication between medical staff. Ordinary ANC systems generally have the secondary path. Therefore, the Filtered-x algorithm diverges if the phase error of the secondary path model does not satisfy the stability condition - π /2 to π /2. In this paper, we focus on the modified-error feedback ANC system. We have proposed a head-mounted ANC system for realizing clear verbal communication between medical staff. This system consists of microphones and loudspeakers located near the user's ear. This system can realize clear verbal communication because the user's ears are not covered with anything. In the head-mounted ANC system, the microphones are located near the user's ear and the user's voice is consequently picked up at the microphones. Therefore, user's voice is also radiated from the loudspeakers as echo signal. A solution to this problem is to utilize a linear prediction filter. The linear prediction filter can seprate wideband noise from narrowband noise by setting an appropriate delay. In this paper, we examine the appropriate delay to separate speech signal and MR noise and the suppression ability of echo signal through computer simulation.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine an automatic parameter adjustment method for audio equalizers using interactive genetic algorithm (IGA). It is very difficult for ordinary users who are not familiar with audio devices to appropriately adjust some parameters of audio equalizers. We have already studied an automatic parameter adjustment method for audio equalizers based on user's evaluation. In this method, the gain and sharpness of the peaking filters which are used for audio equalizers are automatically adjusted based on the user's evaluation. In this paper, we examine that the elite section is effective for users through some subjective tests and evaluation time.

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Language：Japanese   Publisher：東京 : 日本音響学会  

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Other Link： https://ndlsearch.ndl.go.jp/books/R000000004-I025130156

Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

We analyze the dynamical behaviors (learning curves) of the active noise control using a statistical-method. Cross-correlation between a primary path and an adaptive filter are treated as macroscopic variable. By taking the correlations between past tap input vectors and the coefficient vector of the adaptive filter into consideration, we obtain simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under the condition in which the tapped-delay line is sufficiently long. In this report, we generalize and relax model restriction of a primary path in an earlier report. The obtained theory quantitatively agrees with the results of computer simulations.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this report, we propose a head-mounted active noise control (ANC) system with speech communication. Magnetic resonance imaging (MRI) device is one of medical equipment which is used to obtain the cross-sectional image of human body and widely used for non-invasive testing in many medical institutions owing to the convenience and safety. Recently, MRI device is also utilized for microwave coagulation therapy using near-real-time MR images. However, MRI device generates a serious noise (over 100 dB SPL). Hence, surgeons and other medical staff (e.g. nurses and anesthetists) are exposed to the large MRI noise for many hours and cannot verbally communicate with each other. We have therefore proposed a head-mounted ANC system for reducing MRI noise and realizing verbal communication under such a loud noise environment. However, MRI device is generally controlled by the operator outside the MRI room. Hence, we need to realize speech communication between inside and outside the room. We therefore try to integrate the speech communication function with the head-mounted ANC system. Concretely, the error microphones and secondary loudspeakers are also used as an interface to realize the speech communication. In this case, the outside voice may be returned through the error microphone, so an audio-integrated ANC system based on the echo cancellation is utilized. Linear prediction filter is also utilized for separating the inside voice from residual noise. In this report, we demonstrate the validity of the proposed ANC system through some noise reduction experiments and subjective assessment tests on phoneme articulation.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose an improvement method of transmission characteristics in acoustic OFDM (orthogonal frequency division multiplexing)using microphone array and adaptive channel shortening. Recently, acoustic OFDM, which uses audible sound waves as an information transmission medium, has been studied. In acoustic OFDM, the transmission characteristics deteriorate owing to acoustic ambient noise and room reverbera- tion because acoustic noise corrupts audio signal embedding information and room reverberation yields intersymbol interference. We therefore propose a combination method of microphone array and adaptive channel shortening in order to improve the transmission characteristics in acoustic OFDM. Some simulation results demonstrate that the proposed combination can improve the transmission characteristics.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a biometrics authentication method using multimodal features in utterance. The multimodal features in utterance consists of lip shape (physical trait), lip motion pattern and voice pattern(behavioral trait). Therefore, the proposed method can be constructed with only a camera extracting lip area and voice without special equipment like other personal authentication methods. Moreover, the utterance phrase itself has a role of a key function by setting up an utterance phrase arbitrarily, and then the robustness of the authentication increases according to the phrase recognition which can reject an imposter with the feature similar to a registrant. In the proposed method, a lip shape and voice features are extracted with edge or texture based feature in a lip image and voice based pitch or spectrum envelope. Experimental results demonstrate the effectiveness of the proposed method.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine the effect of spatial relationship of acoustic holes to acoustic characteristics by using Finite Element Method(FEM). We quantitatively clarify the effect on acoustic characteristics from spatial relationship between acoustic holes using the acoustic parameter estimation method we developed. A new design guideline for compact acoustic reproduction systems are provided through examining the relationship between simulation and the corresponding experimental results.

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

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We analyze the dynamical behaviors (learning curves) of active noise control with FXLMS algorithm using statistical mechanical method. The cross-correlation between an unknown system and an adaptive filter and autocorrelation of the adaptive filter are treated as the macroscopic variables. We obtain the simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables under the conditions in which the reference signal is white and the tapped-delay line is long. We analytically solve the equations. Neither the independence assumption, the sinusoidal input assumption, the small step-size condition, nor the few-taps assumption is used. We discuss the systematic behaviors and adaptation rate using the derived theory and compare the theory with the simulation results using the real impulse response data of the primary path measured in the laboratory.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine an automatic parameter adjustment method for audio equalizers using interactive genetic algorithm (IGA). It is very difficult for ordinary users who are not familiar with audio devices to appropriately adjust some parameters of audio equalizers. We have already studied an automatic parameter adjustment method for audio equalizers based on user's evaluation. In this method, the gain and sharpness of the peaking filters which are used for audio equalizers are automatically adjusted based on the user's evaluation. In this paper, we examine whether the tournament or the ranking evaluations is appropriate for users through some subjective tests and evaluation time.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Feedback active noise control (ANC) system consists of an error microphone and a secondary source. Even if the noise is generated everywhere, this system can reduce the noise which has periodicity. However, the ANC system can only reduce the noise at the error microphone. Therefore, the ANC system cannot achieve a adequate cancelling performance if we want to reduce the noise at any other different place. In this paper, we examine an ANC system using a virtual microphone, which can reduce the noise at a desired position where the error microphone does not exist, and demonstrate the effectiveness through some simulation and experimental results.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Manufacturing equipment generates a serious noise (over 90 dB) in the factory. Hence, workers are exposed to the industrial noise for many hours and cannot communicate with each other. We therefore attempt to implement an active noise control (ANC) system in order to reduce the industrial noise. It is important to raise the correlation between a reference signal and an error signal in a feedforward ANC system for the noise in the factory. In order to raise the correlation, the distance between the reference and the error microphones must be shortened, or many microphones must be utilized. However, if many microphones are used, the computational complexity increases. Hence, we apply a parametric array loudspeaker as the control (secondary) source. In this case, the increase of sound pressure level in the uncontrolled area may be avoided because the parametric array loudspeaker has the feature of superdirectivity. In this paper, therefore, we examine the ANC system using the parametric array loudspeaker for the noise in the factory.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a biometrics authentication method using multimodal features in utterance. The multimodal features in utterance consists of lip shape (physical trait), lip motion pattern and voice pattern(behavioral trait). Therefore, the proposed method can be constructed with only a camera extracting lip area and voice without special equipment like other personal authentication methods. Moreover, the utterance phrase itself has a role of a key function by setting up an utterance phrase arbitrarily, and then the robustness of the authentication increases according to the phrase recognition which can reject an imposter with the feature similar to a registrant. In the proposed method, a lip shape and voice features are extracted with edge or texture based feature in a lip image and voice based pitch or spectrum envelope. Experimental results demonstrate the effectiveness of the proposed method.

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Language：Japanese   Publisher：関西大学先端科学技術推進機構  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Feedback active noise control (ANC) system which is effective for narrowband noise can get an input signal and monitor the noise reduction performance with only an error microphone. Ordinary ANC systems generally have the secondary path following the noise control filter. Therefore, the noise control filter should be updated by the Filtered-x algorithm. However, the Filtered-x algorithm diverges if the phase error of the secondary path model does not satisfy the stability condition -π/2 to π/2. In this paper, we focus on the modified error feedback ANC system which can update the noise control filter by ordinary adaptive algorithm like NLMS instead of the Filtered-x algorithm. However, the modified error feedback ANC system has a possibility of divergence because the disturbance of broadband noise such as background noise always corrupts the input signal for the system. In this paper, we propose a modified error feedback ANC system using a linear prediction filter in order to suppress the broadband disturbance. When we use the MR noise as the noise targeted for control, noise reduction experiments show that the proposed system is superior to the conventional system on the noise attenuation level while maintaining the stability.

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Language：Japanese   Publisher：関西大学先端科学技術推進機構  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a biometrics authentication method using features in utterance. The features in utterance consists of lip shape (physical trait), lip motion pattern and voice pattern(behavioral trait). Therefore, the proposed method is a multimodal authentication method which consists of an image and a speech processing part. The proposed method can be constructed with only a camera extracting lip area and voice without special equipment like other personal authentication methods. Moreover, the utterance phrase itself has a role of a key function by setting up an utterance phrase arbitrarily, and then the robustness of the authentication increases. Experimental results demonstrate the effectiveness of the proposed method.

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Language：Japanese   Publisher：The Physical Society of Japan (JPS)  

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

We analyze the dynamical behaviors (learning curves) of active noise control with Filtered-X LMS algorithm using statistical mechanical method. Direction cosines among coefficient vectors of an adaptive filter, its shifted filters, and an unknown system are treated as the macroscopic variables. We obtain the simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables when the tapped-delay line is long. We analytically solve the equations. Neither the independence assumption, nor the sinusoidal input assumption, nor the small step-size condition, nor the few-taps assumption is used. The obtained theory quantitatively agrees with computer simulations even when there is some error on a secondary path estimation or the tapped-delay line of the unknown system is longer than that of the adaptive filter.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

We analyze the dynamical behaviors (learning curves) of active noise control with Filtered-X LMS algorithm using statistical mechanical method. Direction cosines among coefficient vectors of an adaptive filter, its shifted filters, and an unknown system are treated as the macroscopic variables. We obtain the simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables when the tapped-delay line is long. We analytically solve the equations. Neither the independence assumption nor the small step sizes condition are used. The obtained theory quantitatively agrees with computer simulations, regardless of whether there is noises on error microphone.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

We analyze the dynamical behaviors (learning curves) of active noise control with Filtered-X LMS algorithm using statistical mechanical method. Direction cosines among coefficient vectors of an adaptive filter, its shifted filters, and an unknown system are treated as the macroscopic variables. We obtain the simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables when the tapped-delay line is long. We analytically solve the equations. Neither the independence assumption nor the small step sizes condition are used. The obtained theory quantitatively agrees with computer simulations, regardless of whether there is noises on error microphone.

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We analyze the dynamical behaviors (learning curves) of active noise control with Filtered-X LMS algorithm using statistical mechanical method. Direction cosines among coefficient vectors of an adaptive filter, its shifted filters, and an unknown system are treated as the macroscopic variables. We obtain the simultaneous differential equations that describe the dynamical behaviors of the macroscopic variables when the tapped-delay line is long. We analytically solve the equations. Neither the independence assumption nor the small step sizes condition are used. The obtained theory quantitatively agrees with computer simulations, regardless of whether there is an error in the secondary path estimation.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a double-talk-detector using multi-modal information (sound and image). An acoustic echo cancellation is used for hands-free telecommunication and teleconference systems. However, the performance of the acoustic echo cancellation deteriorates according to a double talk where the near-end talker and the far-end talker simultaneously utter. For this problem, the acoustic echo canceller (AEC) using Sub-Adaptive-Filter (Sub-ADF) has been already proposed. However, the double-talk detector cannot detect double-talk situations correctly. Therefore, we propose a double-talk detector using multi-modal information in order to improve the performance of the double-talk detector. The proposed double-talk detector detects a voice activity from image information which is obtained from binarized lip image and acoustic information which is obtained from the correlation between the microphone output and the adaptive filter output. Simulation results demonstrate that the proposed double-talk detector can improve the performance compared with the conventional one.

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In this paper, we propose classifier generation methods for a personal authentication system using lip variation. The personal authentication system using lip variation utilizes lip shape (physical trait) and utterance pattern (behavioral trait), and can be constructed with only a camera extracting lip area without special equipment like other personal authentication. We have already proposed a personal authentication system consisting of classifiers judging individuals. The classifiers are generated with features extracted from devided lip region (cell), and is robust for slight lip variation. However, in the conventional classifier generation methods all features used for authentication are uniformly employed, and the less-useful features consequently affect the authentication. Therefore, we propose novel classifier generation methods weighting each features by using AdaBoost. More useful features affect the authentication by weighting each features. Experimental results demonstrate that the proposed method is effective in the actual authentication system.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we demonstrate the compensation effect of nonlinear distortion of loudspeaker systems by using dynamic distortion method. The swept sine wave is usually used for the verification of the compensation effect of nonlinear distortion. However, the evaluation result is not always corresponding to hearing sense because the signals input to loudspeaker systems are a wideband like music and voice. We therefore use dynamic distortion method by white noise which is a wideband signal. We design two linearization systems which using Volterra filter and Mirror filter which using the linear and the nonlinear parameters of a loudspeaker system estimated by Simulated Annealing(SA), and examine the effectiveness on compensating nonlinear distortions of the loudspeaker system. Experimental results show that the dynamic distortion method has effectivity.

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In this paper, we study a method for analyzing compact acoustic reproduction systems through acoustic equivalent circuit. In recent years, the improvement of sound quality has been demanded as the mobile phones are made smaller and more complicated. However, the design of acoustical structure for mobile phones increases more and more difficulty. Generally, the analysis through acoustic equivalent circuits is used for the design of acoustic systems. However, the conventional acoustic formulae cannot be used for the real acoustic design. We therefore modify the acoustic formulae so that the relative position of acoustic holes can be explained through the acoustic equivalent circuit.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a transmission method to reduce a nonlinear distortion, which is a problem when a OFDM (Orthogonal Frequency Division Multiplexing) is used for RFID (Radio Frequency Identification) with a superregeneration method. The amplification characteristic in the superregeneration method has nonlinearity, various nonlinear component consequently occur when a multicarrier signal passes through it. For this problem, the OFDM-FSK (Frequency Shift Keying) which can reduce the number of carrier waves is effective and reduces the influence of the nonlinear distortion. However, a transmission characteristic deteriorates when the number of FSK increases in the OFDM-FSK. We therefore propose an adaptive OFDM-FSK to reduce the influence of the nonlinear distortion by using different multiple values of FSK. We show that a transmission characteristic can be largely improved by using the proposed method through some computer simulations.

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In this paper, we propose a low complexity solution to the compensation of nonlinear distortion of loudspeaker systems. Nonlinear distortion can, in general, be compensated by a linearization system that uses the Volterra kernel. The main problem with this approach is the huge computational complexity associated with convolving the input signal to the 2nd-order Volterra kernel. The subband adaptive Volterra filter (SBAVF) offers lower computational complexity while maintaining the compensation performance. However, the reduction in computational complexity is modest. Our solution is assigning different tap lengths for each sub-band. Experiments show that it significantly reduces the computational complexity while maintaining comparable compensation performance.

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In this paper, we propose a method for analyzing compact acoustic reproduction systems (e.g. mobile phones) through acoustic equivalent circuit. The conventional acoustic formulae, which relate acoustic parameters in acoustic equivalent circuit to acoustic structure, cannot explain the variation of frequency response according to the relation between holes. We therefore clarify how the variation of the relation between holes works the acoustic parameters through finite element method and modify the acoustic formulae according to analysis results through finite element method. We demonstrate that the modified acoustic formulae could explain actual acoustic phenomena on mobile phones well. -

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we calculate the average of the Volterra kernel in the same production lot on Sounder for the cellular phone and show the effectiveness of compensation of nonlinear distortion using the averaged Volterra kernel. We have compensated nonlinear distortions of a loudspeaker system using the Volterra kernel to the corresponding loudspeaker system. However, it is obviously inefficient that each Volterra kernel is measured for all loudspeaker systems in the same production lot. We therefore examine the difference of nonlinear characteristics in the same production lot on sounder for the cellular phone. Moreover, we calculate the sample-averaged Volterra kernel from some samples in the same production lot and show the effectiveness of compensation of nonlinear distortion using the sample-averaged Volterra kernel.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we study the improvements on convergence speed and system stability of active noise control system using simultaneous perturbation algorithm. The convergence speed could be improved by setting perturbation magnitude to optimal value at each frequency bin. The system stability is also improved by incorporating leaky algorithm. However, the threshold which determines the stability of filter coefficients is needed in the leaky algorithm. In this paper, we clarify the optimal threshold through some experimental results.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we study the improvements on convergence speed and system stability of active noise control system using simultaneous perturbation algorithm. The convergence speed could be improved by setting perturbation magnitude to optimal value at each frequency bin. The system stability is also improved by incorporating leaky algorithm. However, the threshold which determines the stability of filter coefficients is needed in the leaky algorithm. In this paper, we clarify the optimal threshold through some experimental results.

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Data embedding to printed images has become an important issue for several applications. The data retrieval techniques from the data embedded printed images have classified into two classes, i.e. a mobile terminal based method and a server based method. In this paper, we assume an original image to be known and the server based method, and a new method for the data embedding to the printed images is proposed. To increase the number of the embedding data bits in the processing of embedding, we adopt "Walsh code" for diffusion code, which is strong to the interference between diffusion codes. As a result, many number of the data bits can be embedded. This can be also applied to the improvement of the detection process and we propose the embedding of same data to three blocks. This technique gives the tolerance to some distortion and noises. In the detection processing, a taken image is forwarded to the server which holds the original image and the embedded data is detected in the server. As a result, more accurate detection can become possible. Finally, the simulation results of the proposed technique show that it can embed many number of the data bits and obtain the high detection rate of the embedding bits.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Data embedding to printed images has become an important issue for several applications. The data retrieval techniques from the data embedded printed images have classified into two classes, i.e. a mobile terminal based method and a server based method. In this paper, we assume an original image to be known and the server based method, and a new method for the data embedding to the printed images is proposed. To increase the number of the embedding data bits in the processing of embedding, we adopt "Walsh code" for diffusion code, which is strong to the interference between diffusion codes. As a result, many number of the data bits can be embedded. This can be also applied to the improvement of the detection process and we propose the embedding of same data to three blocks. This technique gives the tolerance to some distortion and noises. In the detection processing, a taken image is forwarded to the server which holds the original image and the embedded data is detected in the server. As a result, more accurate detection can become possible. Finally, the simulation results of the proposed technique show that it can embed many number of the data bits and obtain the high detection rate of the embedding bits.

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In this paper, we propose a way to determine the nonlinear parameters of a loudspeaker system by Simulated Annealing (SA) for Mirror filter. The nonlinear parameters determined by the conventional W. Klippel's method sometimes become very large values or imaginary values, and then we have not obtained the enough performance on compensating nonlinear distortions of a loudspeaker system when Mirror filter is designed by using the parameters. In contrast, the proposed method is a way to measure the spectrum of a loudspeaker system on the displacement of a diaphragm and find the nonlinear parameters by SA so as to approach the spectrum. They are consequently determined with accuracy. We design Mirror filter which uses the nonlinear parameters of a loudspeaker system determined by SA, and examine the effectiveness on compensating nonlinear distortions of a loudspeaker system. Experimental results demonstrate that the levels of nonlinear distortions can be reduced in the range of 5[dB] to 15[dB] compared with before compensation in case of using the nonlinear parameters determined by SA.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

This paper proposes an improvement for a data embedding method to printed images based on spread spectrum. This technique is based on DCT and PN sequences. First, the conventional embedding method is introduced. In the conventional method, deformations, such as rotation and scaling, have given a serious defect to the extraction rate. To solve this problem, a new data embedding position and the processing by the resolution of scanner itself are applied. An adaptive control technique of the gain coefficient is also proposed. Finally, the simulation result shows the improvement of the extraction rate by the proposed method and the effectiveness of the proposed method.

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In this paper, we propose an improving method of the convergence speed in the sound reproduction system using the simultaneous perturbation method. In sound reproduction using loudspeakers, the preprocessing filters must be constituted to remove the influence of transfer functions between loudspeakers and control points and crosstalk paths. In the conventional sound reproduction system, however, the reproduced sound is degraded by movements of control points. Although the problem is solved by using the simultaneous perturbation method, this system suffers from the disadvantage of slow convergence. Therefore, we propose an improving method of the convergence speed with the delay control filters which compensate only the delay of the preprocessing filters before and after the movements. We also propose a method which considers distance decay of sound pressure due to the movements. Simulation results demonstrate that the proposed methods can realize reasonable convergence speed.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Active noise control (ANC) system using the simultaneous perturbation method has an excellent feature that the excited noise can be controlled even if the control point is moved, because the secondary path model is unnecessary. However, the conventional perturbation method has a problem that the convergence speed slows as the frequency of noise becomes high. In addition, when the frequency characteristic of the secondary path has dips, the cancellation performance deteriorates since it is not be able to control perturbation magnitude well. Therefore, we propose an updating algorithm considering the frequency characteristic of secondary path and a frequency domain variable perturbation control method. The effectiveness of the proposed methods is demonstrated through simulation results.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose an improving method on the convergence property of the simultoneus perturbation(SP) method with step size parameter control. To realize sound reproduction with loudspeaker, it is necesarry to design the inverse filters to cancel the effect of transfer functions. In real enviroment, it is very hard to design the inverse filters because the transfer functions change easily due to listerner's movement. However, by using the SP method, it is possible to design the inverse filters even if the transfer functions change. However, the SP method has a problem that the convergence speed is slow. We therefore propose a method that calculates step size parameters from frequency-domain RA(reproduction accuracy) to improve the convergence speed.

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This paper proposes a novel transmit diversity system at down-link in TDD/MC-CDMA. In the conventional MC-CDMA transmit diversity system, the orthogonality of the spread code is lost and then the BER performance deteriorates in downlink because the data symbols of multiple users are multiplexed by adding the different weights for each subcarrier from many antennas. The proposed system therefore considers transmit weighting and multiplexing schemes based on careful study of the orthogonality of spreading code. In the proposed scheme, loss of the orthogonality of spread code can be prevented and the transmission performance can be improved by dividing the spreading code into some blocks according to the transmission channel environment and adding the same weight. Moreover, many data symbols can be transmitted simultaneously by increasing the number of divisions of spreading code, and the transmission efficiency can be consequently improved. Simulation results demonstrate that the proposed scheme can improve the BER performance and the transmission efficiency.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we study an ideal frequency response for the design of intra concha headphones. The frequency response is the most important physical characteristic for the design of acoustic systems. However, the ideal frequency response for the design of all kinds of headphones including intra concha headphones has not been clarified yet. We therefore propose the head related transfer functions (HRTFs), which are measured at the point (the distance is 1[m]) in front or 30 degree position of right and left of an ideal (distortionless) loudspeaker in free space, as an ideal frequency response for the design of headphones. We also examine the effectiveness of HRTFs as an ideal frequency response in detail through some listening tests. We examine why the headphones whose frequency response is HRTF have a good sound quality.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

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Active noise control (ANC) system using the simultaneous perturbation method has an excellent feature that the excited noise can be controled even if the control point is moved, because the secondary path is unnecessary. However, since the convergence speed is slow, this system cannot track to quick movement of the control point. In this paper, We propose a variable parameters method according to the convergence situation. The variable parameters are magnitude perturbation, step-size and so on, and are very important in simultaneous perturbation method. the effetiveness of the proposed method is demonstrated through simulation and actual experiment result.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we propose a novel sound reproduction system using the perturbation method. The multichannel sound reproduction systems need the inverse filters for cancelling the effect of room transfer functions. However, if a lisetener moves, room transfer functions fluctuate in real acoustic enviroment. Hence, the reproduced sound is degraded by the fluctuation of room transfer functions. To solve this problem, we use the perturbation method, which can work stably under the fluctation of room transfer functions. Simulation results demonstrate that the proposed method can track the fluctation of room transfer functions.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine a sub adaptive digital filter(Sub-ADF) which plays the auxiliary role for an adaptive digital filter(ADF) in acoustic echo cancellation. A feature of Sub-ADF is to be able to estimate the updating state of ADF. Sub-ADF can consequently control the step-size parameter of ADF optimally. In this paper, we also propose an echo path change detector which uses variations of ADF coefficients. As a result, the proposed method is robust to echo path change and can improve the ERLE from 10 to 15 [dB] compared with some conventional methods. Hence, the proposed method can realize superior convergence performance to the conventional ones under bad environments such as the double-talk.

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In this paper, we show acoustic design results for mobile phones using the acoustic design support software for compact acoustic systems we have constructed and examine the effectiveness. Some problems on acoustic quality of mobile phones are now due to the leak of sound and the difficulty of acoustic design. We have therefore constructed acoustic design support software for compact acoustic systems. We implement design algorithms fitting for the actual design situation and based on the design method for acoustic parameters using genetic algorithms we have already proposed in this software. This software has another role on man-machine interface aimed at acoustic design support for acoustic engineers. This paper explains the constructed design support software and demonstrates the effectiveness by examining some design results on various design conditions (diaphragms, size and structure of acoustic elements, and artificial ear).

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we examine a sub adaptive digital filter(Sub-ADF) which plays the auxiliary role for an adaptive digital filter(ADF) in acoustic echo cancellation. A feature of Sub-ADF is to be able to estimate the updating state of ADF. Sub-ADF can consequently control the step-size parameter of ADF optimally. In this paper, we also propose an echo path change detector which uses variations of ADF coefficients. As a result, the proposed method is robust to echo path change and can improve the ERLE from 10 to 15 [dB] compared with some conventional methods.nee, the proposed method can realize superior convergence performance to the conventional ones under bad environments such as the double-talk.

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MC-CDMA adaptive modulation scheme has been studied as a mobile communication system that realizes a high quality and high speed transmission. Moreover, the multimedia service has been developed in the next generation mobile communications as the video is transmitted in real time in addition to the voice service. Ever higher speed transmission and efficient use of frequency are demanded to realize these multimedia communi- cations. Therefore this paper proposes a MC-CDMA adaptive modulation scheme with selects modulation scheme and spreading factor according to the transmission channel environment as a system that realizes ever higher speed transmission. In the conventional MC-CDMA adaptive modulation schemes, the received power is different in each subcarrier because the arrangement of subcarriers is sensitive to the frequency selective fading, therefore high speed transmission cannot be realized. On the other hand, the proposed scheme can improve the transmission efficiency compared with the conventional scheme because the number of data symbols transmitted at a time by the same frequency is increased. Simulation results demonstrate that the proposed scheme can improve the transmission efficiency without deteriorating the BER performance.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

The proposed transmission system therefore assigns each sub-band data suitable weights according to the significance by using adaptive modulation system so that bit errors are concentrated in high-frequency sub-bands which are little significant on image reconstruction. Concretely speaking, significant sub-band data are transmitted by a robust modulation for noises so that the bit errors of the significant sub-bands are reduced. On the other hand, the other sub-band data are transmitted by the high efficient modulation in order to increase transmission rates. A four-division system divides a frame data into four parts and allocates a robust modulation in order from the head. A threshold system changes modulation system for frame data adaptively. Furthermore, audio data are transmitted with high quality by a robust modulation. These proposed system adaptively perform the above process for video and audio data. Moreover, these proposed systems are effective in progressive order of LRCP(Layer-resolution level-component-position) and RLCP(Resolution level-layer-component-position) on JPEG2000. Simulation results demonstrate that these proposed systems can improve image quality compared with the conventional transmission under indoor environment in DS-CDMA.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we study an ideal frequency response for the design of intra concha headphones. The frequency response is the most important physical characteristic for the design of acoustic systems. However, the ideal frequency response for the design of all kinds of headphones including intra concha headphones has not been clarified yet. We therefore propose the head related transfer functions (HRTFs), which are measured at the point (the distance is 1[m]) in front or 30 degree position of right and left of an ideal (distortionless) loudspeaker in free space, as an ideal frequency response for the design of headphones. We also examine the effectiveness of HRTFs as an ideal frequency responsein in detail through some listening tests and the relation between ideal frequency response and sound quality.

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In this paper, we propose a composition of acoustic echo canceller with a noise suppression circuit using linear prediction. In the acoustic echo canceller, the control of the step-size parameter(SSP) is essential to cope with the double-talk, echo path changes, and so on. Therefore, we use the SSP control technique based on the desired ERLE. The technique controls the update of adaptive digital filter (ADF) according to the power of input and noisesignal. However, the technique causes a decrease in the convergence speed under high noise environments becausethe frequency updating the ADF decreases. Hence, the proposed method suppresses the noise by single microphoneand can consequently improve about 5-10dB in SNR. As a result, the proposed method can improve call qualityand realize high accurate estimation and increase in the convergence speed.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

This paper proposes a MC-CDMA adaptive modulation scheme to select subcarriers according to the transmission channel environment. In the conventional MC-CDMA adaptive modulation schemes, the received power is different in each subcarrier because the arangement of subcarriers is sensitive to the frequency selective fading, and then high efficient modulation is inevitably selected. Moreover, the subcarriers with low received power lead to the deterioration of BER performance because the selected modulation scheme cannot satisfy the desired transmission quality. On the other hand, the proposed scheme can increase the average received power because the subcarrier of high received powers and that of low one are combined. As a result, high efficient modulation can be selected and then the deterioration of BER performance can be suppressed. Simulation results demonstrate that the proposed scheme can improve the transmission efficiency and increase about 5dB in BER performance.

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There is a multi-channel ANC system as a technique of controlling noise in larger space. However, since this system needs many error microphone to pick up noise signals, they give a correlation between each reference signal and the convergence speed of noise control filters becomes slow. In this paper, we discuss the effect of transfer function from noise sources to reference microphones and verify the effectiveness of ANC system using the perturbation method with the correlation removal filter through computer simulation and actual systems. We also describe the number of tap required for a correlation removal filter. Further, we use the frequency domain time difference simultaneous perturbation method (FDTDSP method) which have been already proposed to update the noise control filters.

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In this paper, we propose a sub adaptive digital filter (Sub-ADF) to control the step-size parameter (SSP) for acoustic echo cancellers. In the acoustic echo canceller, it is desirable to make adjustment of an adaptive filter small as the estimate progresses. However, it is necessary to bring the updating forward at the time of echo path changing, and to stop updating at the time of a double talk. Therefore the proposed method uses a sub-ADF in order to avoid the problem of the double-talk. The sub-ADF is not influenced by the double-talk and can approximate the present update state so that the proposed method can update the filter coefficients by the optimal step-size parameter. Hence, the proposed method can realize superior convergence performance to the conventional ones under bad environments such as the double-talk.

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Language：Japanese   Publisher：東京 : 電子情報通信学会  

コレクション : 国立国会図書館デジタルコレクション > デジタル化資料 > 雑誌

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

In this paper, we present the effectiveness of eliminating nonlinear distortions of loudspeaker systems for high sampling audio using Volterra filter. We have presented some identification methods and some compensation methods of nonlinear distortion of loudspeaker systems for audio, desktop and mobile phones. By using these compensation methods, the nonlinear distortion of loudspeaker systems for high sampling audio can be compensated. In this paper, we aim at the nonlinear distortion produced by multi-sinusoidal waves on unaudible frequency band. We therefore present the effectiveness of eliminating nonlinear distortions of loudspeaker systems for high sampling audio through some experiments of eliminating the 2nd-order nonlinear distortions. Experimental results show that the levels of the 2nd-order nonlinear distortions can be reduced in the range of 5[dB] to 15[dB] compared with before elimination over the whole frequency band.

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This paper proposes a novel updating method which updates some adaptive filters synchronously in the system which consists of adaptive antenna array (AAA), interference canceller (1C), and equalizer (LE). The conventional updating method updates each adaptive filter independently using one error signal. However, since each adaptive filter searches to the optimal solution independently, the conventional method cannot search the whole optimal solution efficiently. Therefore, we propose a synchronous updating method in which the coefficients of each adaptive filter are integrated into one coefficient vector. The proposed updating method can search the whole optimal solution efficiently. Simulation results demonstrate that the proposed method can improve the convergence speed and realize higher BER performance than the conventional one.