A robust multichannel sound reproduction system that utilizes the relationship between the width of the actual control area and the control frequency of the control points is proposed. The reproduction accuracy of a conventional sound reproduction system is reduced by room environment variations when fixed inverse filter coefficients are used. This tendency becomes more significant when control points are arranged more closely. To resolve this problem, the frequency control band at every control point is switched to avoid degrading the reproduced sound in low frequencies, so the pass band range of the control points at both ears is only high-range. That of the other control points is the entire control range. Numerical simulation with real environmental data showed that improvement of the reproduction accuracy is about 6.1 dB on average, even with a temperature fluctuation of 5C as an environmental variation in the listening room.

We describe a method of compensating temperature fluctuation by a linear-time-warping processing in a sound reproduction system. This technique is applied to impulse responses of room transfer functions, to achieve a high-quality sound reproduction system, particularly one that treats high-frequency components. First, the impulse responses are measured before and after temperature fluctuation, and the former are converted to the latter by the proposed process. Next, we design inverse filters for the system, and evaluate the improvement of the reproduction accuracy and spectrum distortion. By the compensation method, we can improve the reproduction accuracy at any frequency. Moreover, we propose an adaptive algorithm for the estimation of a suitable warping ratio, using the observed signal of reproduced sound obtained at only one control point. Using the proposed algorithm, we can improve the reproduction accuracy at each control point by about 14 dB, in which a difference in temperature is 1.4.

In this paper, we describe a new interface for a barge-in free spoken dialogue system combining multichannel sound field control and beamforming, in which the response sound from the system can be canceled out at the microphone points. The conventional method inhibits a user from moving because the system forces the user to stay at a fixed position where the response sound is reproduced. However, since the proposed method does not set control points for the reproduction of the response sound to the user, the user is allowed to move. Furthermore, the relaxation of strict reproduction for the response sound enables us to design a stable system with fewer loudspeakers than those used in the conventional method. The proposed method shows a higher performance in speech recognition experiments.

To achieve a sound field reproduction system, it is important to design multichannel inverse filters which cancel the effects of room transfer functions. The design method in the frequency domain based on the least-norm solution (LNS) requires less memory and less calculation than the design method in the time domain. However, the LNS method cannot guarantee the causality or stability of the filters. In this paper, a design method of a time-domain inverse filter using iterative processing in the frequency domain for multichannel sound field reproduction is proposed, and the result of numerical analysis is described. The proposed method can decrease the squared error of every control point by 3-12 dB. Furthermore, the sound reproduced by this method attains over 13 dB improvement in the segmental signal-noise ratio (SNR) compared with that designed by the LNS method for real environment impulse responses.

In this paper, we propose a new on-line adaptive relaxation algorithm for an inverse filter in a multichannel sound reproduction system. The fluctuation of room transfer functions degrades reproduced sound in conventional sound reproduction systems in which the coefficients of the inverse filter are fixed. In order to resolve this problem, an iterative relaxation algorithm for an inverse filter performed by truncated singular value decomposition (adaptive TSVD) has been proposed. However, it is difficult to apply this method within the time duration of the sound of speech or music in the original signals. Therefore, we extend adaptive TSVD to an on-line-type algorithm based on the observed signal at only one control point, normalizing the observed signal with the original sound. The result of the simulation using real environmental data reveals that the proposed method can always carry out the relaxation process against acoustic fluctuation, for any time duration. Also, subjective evaluation in the real acoustic environment indicates that the sound quality improves without degrading the localization.

In this paper, we propose a computationally efficient method of compensating temperature for the transaural stereo. The conventional method can be used to estimate the change in impulse responses caused by the fluctuation of temperature with high accuracy. However, the large amount of computation required makes real-time implementation difficult. Focusing on the fact that the amount of compensation depends on the length of the impulse response, we reduce the computation required by segmenting the impulse response. We segment the impulse responses in the time domain and estimate the effect of temperature fluctuation for each of the segments. By joining the processed segments, we obtain the compensated impulse response of the whole length. Experimental results show that the proposed method can reduce the computation required by a factor of nine without degradation of the accuracy.