Shoichi Koyama

Reproducing Virtual Sound Sources in Front of a Loudspeaker Array Using Inverse Wave Propagator

It has been possible to reproduce point sound sources between listeners and a loudspeaker array by using the focused-source method. However, this method requires physical parameters of the sound sources to be reproduced, such as source positions, directions, and original signals. This fact makes it difficult to apply the method to real-time reproduction systems because decomposing received signals into such parameters is not a trivial task. This paper proposes a method for recreating virtual sound sources in front of a planar or linear loudspeaker array. The method is based on wave field synthesis but extended to include the inverse wave propagator often used in acoustical holography. Virtual sound sources can be placed between listeners and a loudspeaker array even when the received signals of a microphone array equally aligned with the loudspeaker array are only known. Numerical simulation results are presented to compare the proposed and focused-source methods. A system was constructed consisting of linear microphone and loudspeaker arrays and measurement experiments were conducted in an anechoic room. When comparing the sound field reproduced using the proposed method with that using the focused-source method, it was found that the proposed method could reproduce the sound field at almost the same accuracy.

Design of transform filter for sound field reproduction using microphone array and loudspeaker array

In this paper, we propose a novel method of sound field reproduction using a microphone array and loudspeaker array. Our objective is to obtain the driving signal of a planar or linear loudspeaker array only from the sound pressure distribution acquired by the planar or linear microphone array. In this study, we derive a formulation of the transform from the received signals of the microphone array to the driving signals of the loudspeaker array. The transform is achieved as a mean of a filter in a spatio-temporal frequency domain. Numerical simulation results are presented to compare the proposed method with the method based on the conventional least square algorithm. The reproduction accuracies were found to be almost the same, however, the filter size and amount of calculation required for the proposed method were much smaller than those for the least square algorithm based one.

Design of transform filter for reproducing arbitrarily shifted sound field using phase-shift of spatio-temporal frequency

For real-time sound field transmission systems from a far-end to a near-end, the driving signals of a loudspeaker array at the near-end need to be calculated by using only received signals obtained by a microphone array at the far-end. Additionally, having the capability to control the location of the sound field to be reproduced in order to adjust it to the visual images is advantageous. The goal of this study was to develop a method to transform received signals of a microphone array into driving signals of a loudspeaker array in order to reproduce arbitrary shifted sound fields. We analytically derive a transform filter in the spatio-temporal frequency domain. The location of the sound field to be reproduced is controllable only by phase-shift of the transform filter. The proposed method was found to be computationally efficient compared to the conventional method based on a least squares algorithm, and numerical simulation results indicated that reproduction accuracies were almost the same in both methods.