This paper introduces the multiple signal classification (MUSIC) method that utilizes the transfer characteristics of microphones located at the same place, namely aggregated microphones. The conventional microphone array realizes a sound localization system according to the differences in the arrival time, phase shift, and the level of the sound wave among each microphone. Therefore, it is difficult to miniaturize the microphone array. The objective of our research is to build a reliable miniaturized sound localization system using aggregated microphones. In this paper, we describe a sound system with N microphones. We then show that the microphone array system and the proposed aggregated microphone system can be described in the same framework. We apply the multiple signal classification to the method that utilizes the transfer characteristics of the microphones placed at a same location and compare the proposed method with the microphone array. In the proposed method, all microphones are placed at the same place. Hence, it is easy to miniaturize the system. This feature is considered to be useful for practical applications. The experimental results obtained in an ordinary room are shown to verify the validity of the measurement.

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 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.

The actual sound environment system exhibits various types of linear and non-linear characteristics, and it often contains an unknown structure. Furthermore, the observations in the sound environment are often in the level-quantized form. In this paper, a method for estimating the specific signal for stochastic systems with unknown structure and the quantized observation is proposed by introducing a system model of the conditional probability type. The effectiveness of the proposed theoretical method is confirmed by applying it to the actual problem of psychological evaluation for the sound environment.

Two methods for hotspot generation using multiple sources, known as time-delay (TD) method and maximum-control-gain (MCG) method are investigated in the two typical acoustical fields, namely, the free field and a rectangular room. Based on the theoretical analysis and simulations, strategies are developed according to the sound field where the target region is defined. In the free field, the MCG method can be used if the performance in terms of control gain is the priority for an optimal control, whereas the TD method is more preferable if the simplicity of implementation is the first consideration. In a room environment, if a target region is defined in the near field where the direct sound dominates, the TD method is still effective. However, in the far field where the reverberant sound prevails, only the MCG method is applicable. The near field/far field can be roughly separated according to the critical distance from the sources in the room.

An algorithm for the sound projection using multiple sources is presented. The source strength vector is obtained by using a fast estimation approach instead of the conventional eigenvalue decomposition (EVD) method. The computation load is therefore greatly reduced, which makes the algorithm more efficient in practical applications.

The discovery of the Turbo codes has driven research on the creation of new signal detection concepts that are, in general, referred to as the Turbo approach. Recently, this approach has made a drastic change in creating signal detection techniques and algorithms such as equalization of inter-symbol interference (ISI) experienced by broadband single carrier signaling over mobile radio channels. A goal of this paper is to provide readers with broad views and knowledge of the Turbo concept-based Multiple-Input Multiple-Output (MIMO) signal transmission techniques. How the techniques have been developed in various applications and how they perform in real-field environments are introduced.

This paper describes a new interface for a barge-in free spoken dialogue system combining an adaptive sound field control and a microphone array. In order to actualize robustness against the change of transfer functions due to the various interferences, the barge-in free spoken dialogue system which uses sound field control and a microphone array has been proposed by one of the authors. However, this method cannot follow the change of transfer functions because the method consists of fixed filters. To solve the problem, we introduce a new adaptive sound field control that follows the change of transfer functions.

An interactive 3-D sound processing system and its implementation is described, which is to provide virtual auditory environments to listeners. While conventional 3-D sound processing systems require high performance workstations or large DSP arrays, the proposed system is reduced in hardware size for practical applications. The proposed system is implemented using a prevailing IBM-compatible PC and a single DSP. Since the organization of the proposed system is independent of implementation details such as operation precision and number of audio tracks, the proposed system can be ported to various hardware entities. In addition, an easy-to-use user interface is also implemented on PC software for realtime input of 3-D sound movement. Owing to these features, the presented system is valuable as a prototype for various implementation of 3-D sound processing systems, while the current implementation is useful as a 3-D sound content production system.

A directional audible sound can be generated by amplitude-modulated (AM) into ultrasound wave from a parametric array. To synthesize audio signals produced by the self-demodulation effect of the AM sound wave, a quasi-linear analytical solution, which describes the nonlinear wave propagation, is developed for fast numerical evaluation. The radiated sound field is expressed as the superposition of Gaussian Beams. Numerical results are presented for a rectangular parametric loudspeaker, which are in good agreement with the experimental data published previously.

An ultra wide band channel sounder has been developed and has attained the time delay resolution of 0.5 ns which enables the propagation path discrimination in indoor wireless propagation environments as well as the direction-of-arrival measurements by power delay profile measurements.

In a two-microphone approach, interchannel differences in time (ICTD) and interchannel differences in sound level (ICLD) have generally been used for sound source localization. But those cues are not effective for vertical localization in the median plane (direct front). For that purpose, spectral cues based on features of head-related transfer functions (HRTF) have been investigated, but they are not robust enough against signal variations and environmental noise. In this paper, we use a "profile" as a cue while using a combination of reflectors specially designed for vertical localization. The observed sound is converted into a profile containing information about reflections as well as ICTD and ICLD data. The observed profile is decomposed into signal and noise by using template profiles associated with sound source locations. The template minimizing the residual of the decomposition gives the estimated sound source location. Experiments show this method can correctly provide a rough estimate of the vertical location even in a noisy environment.

A differential electromagnetic transducer (DET) was implemented using micro electro mechanical system (MEMS) technology for use in an implantable middle ear (IME) system. The DET is designed to have good vibration efficiency and structure that can't be interfered by the external environmental magnetic field. In order to preserve the uniform vibration performance, the MEMS technology was introduced to manufacture the elastic membrane using polyimide that is softer than silicon. Using the finite element analysis (FEA), vibration characteristics are simulated and designed so that the resonance frequency of the membrane is closed to that of the middle ear. The results of the vibration experiments of the developed DET showed excellent results. We implemented the IME system using a DET and implanted it into a dog. This showed the IME system performed well in a living body.

This paper describes a method of analyzing musical sound using a self-organizing map. To take compound factors into account, energy spectra whose frequency ranges were based on the psycho-acoustic experiments were used as input data. Results for music compact discs confirmed that our method could effectively display the positioning and relationships among musical sounds on a map.

This letter outlines a scheme to produce a wider robust bandwidth, with better approximations to the perfect reproduction of pre-recorded acoustic signals. Multi-parameter inverse filtering method is proposed in the virtual sound imaging system for improving the robustness performance. The superiority of this new type of inverse filter is demonstrated on a 3-speaker system.

Heart sound is an acoustic wave generated by the mechanical movement of the heart and blood flow, and is a complicated, non-stationary signal composed of many signal sources. It can be divided into normal heart sounds and heart murmurs. Murmurs are abnormal signals that appear over wider ranges of frequency than normal heart sounds. They are generated at random spots in the whole period of heart sounds. The recognition of heart sounds is to differentiate heart murmurs through patterns that appear according to the generation time of murmurs. In this paper, a group of heart sounds was classified into normal heart sounds, pre-systolic murmurs, early systolic murmurs, late systolic murmurs, early diastolic murmurs, and continuous murmurs. The suggested algorithm was standardized by re-sampling and then added as an input to the neural network through wavelet transform. The neural network used Error Back - Propagation algorithm, which is a representative learning method, and controlled the number of hidden layers and the learning rate for optimal construction of networks. As a result of recognition, the suggested algorithm obtained a higher recognition rate than that of existing research methods. The best result was obtained with the average of 88% of the recognition rate when it consisted of 15 hidden layers. The suggested algorithm was considered effective for the recognition of automatic diagnosis of heart sound recognition programs.

Recently, information display equipment such as a navigation system has often come to be installed in a vehicle, and a variety of useful information has been offered to the driver by voice and images while driving. The necessity of improving safety when the driver receives such information has come to be stressed. As one of the means of solving this problem, we can develop a system that presents the driving and road conditions information such as a lane changing car to the driver by using a warning sound. The purpose of our study is to clarify the effectiveness of an auditory display that uses spatial sounds on such a system. An experiment for measuring the driver's reaction time and eye movements to LED lighting during actual driving has been carried out to investigate whether the spatial sound can quicken the driver's operation and decrease human error. We evaluated the effectiveness by two measures, average reaction time and the number of largely delayed reactions. We considered that the average reaction time corresponds to the quickness of the driver's operation, and the number of largely delayed reactions corresponds to the probability of human error. As a result of the experiment, the use of directional sound clearly showed better performance than the use of monaural sound and no sound in the number of largely delayed reactions. Moreover, we analyzed the factors involved in delay of the reaction by the results of eye movement measurements. Consequently, it has been found that directional sound can decrease the number of the largely delayed reactions, which lead to an accident during actual driving.

In loudspeaker-based 3D audio systems, there are some acoustic crosstalk cancellation methods to enlarge the 'sweet spot' around a fixed listener position. However, these methods have common defect that most of them can be applied only to the specific narrow frequency band. In this letter, we propose the more robust acoustic crosstalk cancellation method so that we can cancel the crosstalk signal in far wider frequency band and enlarge 'sweet spot. ' For this goal, we apply a sum and difference filter to the conventional three loudspeaker-based 3D audio system.

A sound localization method based on the adaptive estimation of inverse Ear Canal Transfer Functions (ECTFs) using a stereo earphone-microphone combination is proposed. This method can adaptively obtain the individual's transfer functions to fit the listener in real-time. We evaluate our sound localization method by studying the relationship between the estimation error of inverse ECTFs and the auditory sound localization score perceived by several listener. As a result, we clarified that the estimation error required of inverse ECTFs are less than -10 dB. In addition, we describe two adaptive inverse filtering methods in order to realize real-time signal processing implementation using affine projection algorithm and discusses the convergence time of an adaptive inverse filter to determine the initial value. It is clarified that method 2 based on copy weights with initial value is more effective than method 1 with filtered-x algorithm, in terms of convergence, if the initial value is the average of many listeners' impulse responses for our sound localization method.

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.