A new real-time sound field transmission system is presented. To construct this system, a large listening area needs to be reproduced at not less than a constant height. Additionally, the driving signals of the loudspeakers should be obtained only from received signals of microphones. Wave field reconstruction (WFR) filtering for linear arrays of microphones and loudspeakers is considered to be suitable for this kind of system. An experimental system was developed to show the feasibility of real-time sound field transmission using the WFR filter. Experiments to measure the reproduced sound field and a subjective listening test of sound localization were conducted to evaluate the proposed system. Although the reproduced sound field included several artifacts such as spatial aliasing and faster amplitude decay, the experimental results indicated that the proposed system was able to provide sound localization accuracy for virtual sound sources comparable to that for real sound sources in a large listening area.

Sound field reproduction systems seek to realistically convey 3D spatial audio by re-creating the sound pressure inside a region enclosing the listener. High-order Ambisonics (HOA), a sound field reproduction technology, is notable for defining a scalable encoding format that characterizes the sound field in a system-independent way. Sound fields sampled with a particular microphone array and encoded into the HOA format can be reproduced using any sound presentation device, typically a loudspeaker array, by using a HOA decoder. The HOA encoding format is based on the spherical harmonic decomposition; this makes it easier to design a decoder for large arrays of loudspeakers uniformly distributed over all directions. In practice, it is seldom possible to cover all directions with loudspeakers placed at regular angular intervals. An irregular array, one where the angular separation between adjacent loudspeakers is not constant, does not perform as well as a regular one when reproducing HOA due to the uneven sampling of the spherical harmonics. This paper briefly introduces the techniques used in HOA and advances a new approach to design HOA decoders for irregular loudspeaker arrays. The main difference between conventional methods and our proposal is the use of a new error metric: the radial derivative of the reconstruction error. Minimizing this metric leads to a smooth reproduction, accurate over a larger region than that achieved by conventional HOA decoders. We evaluate our proposal using the computer simulation of two 115-channel loudspeaker arrays: a regular and an irregular one. We find that our proposal results in a larger listening region when used to decode HOA for reproduction using the irregular array. On the other hand, applying our method matches the high-quality reproduction that can be attained with the regular array and conventional HOA decoders.

In this paper, we propose a new system for controlling radiated sound directivity. The proposed system artificially induces a bending vibration on a planar diaphragm by vibrating it artificially using multiple vibrators. Because the bending vibration in this case is determined by not one but all of the accelerated vibrations, the vibration of the diaphragm can be controlled by modulating the accelerated vibration waveforms relatively for each frequency. As a consequence, the directivity of the radiated sound is also varied. To investigate the feasibility of this system, we constructed a prototype that has for a diaphragm a circular plate-one of the most typical shapes considered for discussing plate vibration-and three vibrators. The measurement data showed visually that with this system, surface vibration and sound directivity change depending on the phases of the accelerated vibrations.

Although it is very important to conduct listening tests when constructing a practical sound field reproduction system based on wave field synthesis, listening tests are very expensive. A localization model of synthesized sound images that predicts the results of listening tests is proposed. This model reduces the costs of constructing a reproduction system because it makes it possible to omit the listening tests. The proposed model uses the precedence effect and predicts the direction of synthesized sound images based on the inter-aural time difference. A comparison of the results predicted by the proposed model and the localized results of listening tests shows that the model accurately predicts the localized results.

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 order to control a sound field using multiple sources and microphones, we must choose the optimum values of parameters such as the numbers of sources and microphones, the location of the sources and the microphones and the filter tap length. Because there is a huge number of possible combinations of these conditions, the boundary surface control principle can be useful as a basis of a design method of such a system. In this paper, a design method of sound field reproduction and active noise control based on the BSC principle are described and several example of its application are presented.

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.