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.

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.

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.

This paper describes automatic generation of speech synthesis rules which predict a stress level for each bunsetsu in long noun phrases. The rules are inductively inferred from a lot of speech data by using two kinds of tree-based methods, the conventional decision tree and the SBR-tree methods. The rule sets automatically generated by two methods have almost the same performance and decrease the prediction error to about 14 Hz from 23 Hz of the accent component value. The rate of the correct reproduction of the change for adjacent bunsetsu pairs is also used as a measure for evaluating the generated rule sets and they correctly reproduce the change of about 80%. The effectiveness of the rule sets is verified through the listening test. And, with regard to the comprehensiveness of the generated rules, the rules by the SBR-tree methods are very compact and easy to human experts to interpret and matches the former studies.

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.

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.

This paper describes a general interface system for speech input and output and a dialog management system, MASCOTS, which is a component of the interface system. The authors designed this interface system, paying attention to its generality; that is, it is not dependent on the problem-solving system it is connected to. The previous version of MASCOTS dealt with the dialog processing only for the speech input based on the SR-plans. We extend MASCOTS to cover the speech output to the user. The revised version of MASCOTS, named MASCOTS II, makes use of topic information given by the topic packet network (TPN) which models the topic transitions in dialogs. Input and output messages are described with the concept representation based on the case structure. For the speech input, prediction of user's utterance is focused and enhanced by using the TPN. The TPN compensates for the shortages of the SR-plan and improves the accuracy of prediction as to stimulus utterances of the user. As the dialog processing in the speech output, MASCOTS II extracts emphatic words and restores missing words to the output message if necessary, e.g., in order to notify the results of speech recognition. The basic mechanisms of the SR-plan and the TPN are shared between the speech input and output processes in MASCOTS II.

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.

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.

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.