An algorithm for extracting fundamental frequencies from duet sounds is proposed. The algorithm is based on an acoustical feature that the temporal fluctuation patterns in frequency an power are similar for harmonic components composing a sound for a single musical note played on a single instrument with a single active vibrating source. The algorithm is applied to the sounds of 153 combinations of pair-notes played by a flute duet and a violin duet. Experimental results show that the zone-wize correct identification rate by pitch name are 98% for the flute duet and 95% for the violin duet in the best cases.

The Multiple input-output INverse/filtering Theorem (MINT) proves that N + 1 inverse filters are necessary to precisely control sound at N points in a space, and gives the minimum orders of such filters. In this paper, we propose the Indefinite MINT Filters (IMFs) for adding one or more control points to the above framework without increasing the number of inverse filters. Although the controllability of the new point is not sufficient, that of the other points is still maintained high enough by the principle of the MINT. In a two point sound control (using two inverse filters), the IMFs could reduce the squared error to the desired sound up to - 10 dB at the second point which is not controlled by the MINT.

In the conventional sound field reproduction system with control of the transfer functions from the source to both ears of a listener, a slight shift of the ears caused by movement of the listener inevitably results in sound localization being different from that expected. In this paper, a method for reproducing a sound field by controlling the transfer function from the source to multiple points (called the "method of multiple-points control" hereafter) is applied to a sound reproduction system with the aim of expanding the area which can be controlled. The system is controlled so that the transfer functions from the input of the system to the multiple points adjacent to the original receiving points have the same desired transfer function. By placing the control points at appropriate intervals, a "zone of equalization" is formed. Based on a computer simulation, the intervals between control points is discussed. The configuration of the loundspeakers for sound reproduction is also discussed.

This paper describes a trial of evaluating the proper characteristics of multiple sound insulatain systems from their output responses contaminated by unknown background noises. The unknown parameters of sound insulation systems are first estimated on the basis of hte linear time series on an intensity scale, describing functionally the input-output relation of the systems. Then, their output probability distributions are predicted when an arbitrary input noise passes through these insulation systems.

This paper proposes a new emphasizing three-dimensional pointing device considering user friendliness and lack of cable clutter. The proposed method utilizes five degrees of freedom via the medium of non-verbal voice of human. That is, the spatial direction of the sound source, the type of the voice phoneme and the tone of the voice phoneme are utilized. The input voice is analyzed regarding the above factors and then taking proper effects as previously defined for human interface. In this paper the estimated spatial direction is used for three-dimensional movement for the virtual object as three degrees of freedom. Both of the type and the tone of the voice phoneme are used for remaining two degrees of freedom. Since vocalization of nonverbal human voice is an everyday task, and the intonation of the voice can be quite easily and intentionally controlled by human vocal ability, the proposed scheme is a new three-dimensional spatial interaction medium. In this sense, this paper realizes a cost-effective and handy nonverbal interface scheme without any artificial wearing materials which might give a physical and psychological fatigue. By using the prototype the authors evaluate the performance of the scheme from both of static and dynamic points of view and show some advantages of look and feel, and then prospect possibilities of the application for the proposed scheme.

In this study, an extraction method of failure sound signal which is strongly contaminated by noise is investigated by genetic algorithm and statistical tests of the frequency domain for the failure diagnosis of machinery. In order to check the extraction accuracy of the failure signal and obtain the optimum extraction of failure signal, the "existing probability Ps (t*k) of failure signal" and statistical information Iqp are defined as the standard indices for evaluation of the extraction results. It has been proven by practical field data and application of the inspection and diagnosis robot that the extraction method discussed in this paper is effective for detection of a failure and distinction of it's origin in the diagnosis of machinery.

In order to estimate elasticity distribution of living soft tissue by ultrasonic pulse-echo method, we developed an algorithm by which we estimate 2-D displacement vector field from two successive rf echo data frames. The algorithm estimates a displacement vector iteratively by matching the phase characteristics of the local regions of two data frames. The estimation process is composed of coarse one and the fine one. In the coarse estimation process, the displacement is estimated by detecting the peak of the 2-D cross-correlation function. In the fine process, the displacement is estimated iteratively by shifting the 2nd frame data so that the phase characteristics matches with that of the 1st frame data. In each iterative step of both processes, the estimated displacement vector field is spatially smoothed. This proposed algorithm exhibits excellent performance in obtaining accurate and smooth distribution of displacement vector which is required to obtain strain distribution and finally shear modulus distribution. We conducted an experiment on an agar phantom which has inhomogeneous shear modulus distribution. Using the proposed method, we obtained 2-D displacement field with reasonable accuracy. We reconstructed a relative shear modulus map using axial strain assuming 1-D stress condition. The reconstructed map using the calculated axial strain through 2-D displacement estimation algorithm was satisfactory, and was clearly superior to the one through 1-D displacement estimation algorithm. The proposed 2-D displacement field estimation algorithm seems to be a versatile and powerful tool to measure strain distribution for the purpose of tissue elasticity estimation under various deformation conditions.

In order to evaluate quantitatively TMJ sound, TMJ sound in normal subject group, CMD patient group A with palpable sounds unknown to them, CMD patient group B with palpable sounds known to them, and CMD patient group C with audible sounds were detected by a contact microphone, and frequency analysis of the power spectra was performed. The power spectra of TMJ sound of normal subject group and patient group A showed patterns with frequency values below 100 Hz, whereas the power spectra of patient groups B and C showed distinctively different patterns with peaks of frequency component exceeding 100 Hz. As regards the cumulative frequency value, the patterns for each group clearly differed from those of other groups; in particular the 80% cumulative frequency value showed the greatest difference. From these results, it is assumed that the 80% cumulative frequency value can be used as an effective indicator for quantitative evaluation of TMJ sound.

This paper proposes a novel phase ambiguity resolver with combining a very low power Viterbi decoder employing a scarce state transition scheme to realize cost effective receivers for the PCM sound broadcasting satellite service. The theoretical analyses on phase decision performance show that the proposed resolver achieves the symbol-by-symbol phase detection and decides correctly phases of the demodulated data even if the bit error probability of 710-2. The resolver also reduces the phase decision time to below 1/1000 of that of the conventional resolver. Furthermore, experimental results of the power consumption estimate that the prototype Viterbi decoder consumes only 60mW at the data rate of 24.576Mbit/s.

A novel pulse neural network model for sound localization has been proposed. Our model is based on the physiological auditory nervous system. Human beings can perceive the sound direction using inter-aural time difference (ILD) and inter-aural level difference (ILD) of two sounds. The model extracts these features using only pulse train information. The model is divided roughly into three sections: preprocessing for input signals; transforming continuous signals to pulse trains; and extracting features. The last section consists of two parts: ITD extractor and ILD extractor. Both extractors are implemented using a pulse neuron model. They have the same network structure, differing only in terms of parameters and arrangements of the pulse neuron model. The pulse neuron model receives pulse trains and outputs a pulse train. Because the pulses have only simple informations, their data structures are very simple and clear. Thus, a strict design is not required for the implementation of the model. These advantages are profitable for realizing this model by hardware. A computer simulation has demonstrated that time and level differences between two signals have been successfully extracted by the model.

A new system of measuring the thickness of thin filn or paper using 40 kHz ultrasonic wave in air is described. The thickness of samples measured is smaller by a factor of sevreal hundreds than the wavelength of sound. Experinents with polymer and metal films and paper are described to demonstrate the measurement possibilities.

In the future, it will be necessary that robot technology or environmental technology has an auditory function of recognizing sound expect for speech. In this letter, we propose a recognition system for the species of birds receiving birdcalls, based on network technology. We show the first step of a recognition system for the species of birds, as an application of a recognition system for environmental sound.

In this study, an expression of the regression relationship with less information loss is concretely derived in the form suitable to the existence of amplitude constraint of the observed data and the prediction of response probability distribution. The effectiveness of the proposed method is confirmed experimentally by applying it to the actual acoustic data.

A new trial of statistical evaluation for an output response of power linear type acoustic systems with nonstationary random input is proposed. The purpose of this study is to predict the output probability distribution function on the basis of a standard type pre-experiment in a laboratoty. The statistical properties like nonstationarity, non-Gamma distribution property and various type linear and non-linear correlations of input signal are reflected in the form of differential operation with respect to distribution parameters. More concretely, the pre-experiment is carried out for a power linear acoustic system excited only by the Gamma distribution type sandard random input. Considering the non-negative random property for the output response of a power linear system, the well-known statistical Laguerre expansion series type probability expression is first employed as the framework of basic probability distribution expression on the output power fluctuation. Then, the objective output probability distribution for a non-stationary case can be easily derived only by successively employing newly introduced differential operators to this basic probability distribution of statistical Laguerre expansion series type. As an application to the actual noise environment, the proposed method is employed for an evaluation problem on the stochastic response probability distribution for an acoustic sound insulation system excited by a nonstationary input noise.

In this paper, an optimized wideband channel sounder designed for measuring the time variant impulse response of outdoor radio channels in the frequency range 1800-2000 MHz is presented. Prior to hardware implementation the system was first modelled on a high performance supercomputer to enable the system designer to optimize the digital signal processing algorithms and the parameters of the hardware components by simulation. It is shown that the proposed measuring system offers a significantly larger amplitude resolution, i.e. dynamic range, than conventional systems applying matched filtering. This is achieved by transmitting digitally generated periodic spread spectrum test signals adjusted to amplifier non-linearities and by applying optimum unbiased estimation instead of matched filtering in the receiver. A further advantage of the hardware implementation of the proposed system compared to conventional systems [5]-[7] is its high flexibility with respect to measuring bandwidth, period of the test signal and sounding rate. The main features of the optimized system are described and first measurement results are presented.

The method of estimating multiple sound source locations based on a neural network algorithm and its performance are described in this paper. An evaluation function is first defined to reflect both properties of sound propagation of spherical wave front and the uniqueness of solution. A neural network is then composed to satisfy the conditions for the above evaluation function. Locations of multiple sources are given as exciting neurons. The proposed method is evaluated and compared with the deterministic method based on the Hyperbolic Method for the case of 8 sources on a square plane of 200m200m. It is found that the solutions are obtained correctly without any pseudo or dropped-out solutions. The proposed method is also applied to another case in which 54 sound sources are composed of 9 sound groups, each of which contains 6 sound sources. The proposed method is found to be effective and sufficient for practical application.

A review is presented of the fundamental principles underlying modern techniques for the active control of acoustic noise. The basic physical principles are first dealt with in the context of the active control of free field radiation and the classical approaches to the problem are briefly discussed. The active control of sound fields in ducts and enclosures is also described and the inherent physical limitations of the technique are emphasised. Modern signal processing methods for realising feedforward control systems are also outlined and least squares formulations are presented which enable performance limits to be established and adaptive algorithms to be derived.

Inverse filters can be designed in order to enhance the accuracy with which signals recorded in a given space can be reproduced in a given listening space. The problem is considered here of the design of an inverse filter matrix which enables K recorded signals to be accurately reproduced at K points in the listening space when transmitted via M loudspeaker channels. The analysis is sufficiently general to incorporate the case when the best (least squares) approximation is sought to the reproduction of K signals at L points in the space when LK. An analysis is presented which demonstrates that the approach suggested by the Multiple-Input/Output Inverse Filtering theorem of Miyoshi and Kaneda can be realised adaptively by using the Multiple Error LMS algorithm of Elliott et al.

This paper describes a new trial of dynamical parameter estimation for the actual room acoustic system, in a practical case when the input excitation is polluted by a background noise in contrast with the usual case when the output observation is polluted. The room acoustic system is first formulated as a discrete time model, by taking into consideration the original standpoint defining the system parameter and the existence of the background noise polluting the input excitation. Then, the recurrence estimation algorithm on a reverberation time of room is dynamically derived from Bayesian viewpoint (based on the statistical information of background noise and instantaneously observed data), which is applicable to the actual situation with the non-Gaussian type sound fluctuation, the non-linear observation, and the input background noise. Finally, the theoretical result is experimentally confirmed by applying it to the actual estimation problem of a reverberation time.

Sound field telecommunication describes a voice communication system, intended to implement a virtual meeting, in which participants at distant sites experience the sensation of sharing a single room for conversation. Binaural synthesis reconstructs the sound propagation pattern of a particular room or environment in the vicinity of each ear, which seems appropriate for a personal multimedia environment. Localization cues in spatial hearing comprise both the sink's transfer function and source attenuation. Sink directional cues are captured by binaural head related transfer functions (HRTFs). Source attenuation is modeled as a frequency-independent function of the direction, dispersion, and distance of the source, capturing sensitivity, amplification, and mutual position. Audio windows, aural analogues of video windows, can be thought of as a user interface to binaural sound presentation for a teleconferencing system. Exocentric representation of audio window entities allows manipulation of all teleconferees in a projected egalitarian medium. We are implementing a system that combines dynamically selected HRTFs with dynamically determined source and sink position, azimuth, focus, and size parameters, controlled via iconic manipulation in a graphical window. With such an interface, users may arrange a virtual conference environment, steering the virtual positions of teleconferees.