This paper proposes a loose part monitoring system (LPMS) design with a signal processing method based on fuzzy logic. Considering fuzzy characteristics of metallic impact waveform due to not only interferences from various types of noises in an operating nuclear power plant but also complex wave propagation paths within a monitored mechanical structure, the proposed LPMS design incorporates the comprehensive relation among impact signal features in the fuzzy rule bases for the purposes of alarm discrimination and impact diagnosis improvement. The impact signal features for the fuzzy rule bases include the rising time, the falling time, and the peak voltage values of the impact signal envelopes. Fuzzy inference results based on the fuzzy membership values of these impact signal features determine the confidence level data for each signal feature. The total integrated confidence level data is used for alarm discrimination and impact diagnosis purposes. Through the perpormance test of the proposed LPMS with mock-up structures and instrumentation facility, test results show that the system is effective in diagnosis of the loose part impact event(i.e., the evaluation of possible impacted area and degree of impact magnitude) as well as in suppressing false alarm generation.

Speech sounds were practically applied to a cochlear filter which was simulated by an electrical transmission line. The amplitude of the basilar membrane displacement was calculated along the length of the cochlea in temporal response. And the envelope of the amplitude according to the length was arranged for each discrete time interval. The resulting time response of the speech sound was then displayed as a color image. Five vowels such as a, e, I, o, u were applied and their results were compared. The whole procedure of the visualization method of the speech sound using the cochlear filter is described in detail. The filter model response to voice is visualized by passing the voice through the cochlear filter model.

A categorization for cluster is necessary when an unsupervised classfication is used for remote sensing image classification. It is desirable that this method is performed automatically, because manual categorization is a highly time consuming process. In this paper, several automatic determination methods were proposed and evaluated. They are four methods. a) maximum number method : which assigns the tharget cluster to the category which occupies the largest area of that cluster b) maximum percentage method : which assigns the target cluster to the category which shows the maximum percentage within the category in that cluster. c) minmun distance method : which assigns the target cluster to the category having minmum distance with that cluster d) element ratio matching method : which assigns local regions to the category having the most similar element ratio of that region From the results of the experiments, it was certified that the result of minimum distance method was almost the same as the result made by a human operator.

In this paper, a new fast voronoi divider for vector quantization (VQ) is introduced, which results from Theorem that the nearest vectors in the sense of minimum mean square error(MMSE) have almost the same mean values of their elements. An improved splitting method for a VQ codebook design using the fast voronoi divider is also presented. Experimental results show that the new method reduces the complexity of training a VQ codebook several times with a high signal to noise ratio(SNR) using an appropriate extensive parameter of codebook.

Matched field localization schemes often show a high sensitivity to acoustic variabilities due to mismatch between assumed and actual environments. In this paper, we focus on the effect of source motion or Doppler on matched field processing (MEP). to accomplish this, MFP is extended to treat a moving source problem with normal mode description of the sound field. the extension involves both the temporally nonstationary and spatially inhomogeneous nature of the sound field generated by a time-harmonic point source moving uniformly in a stratified oceanic waveguide. It is demonstrated that the impact of source motion can be significant to MEP although the velocity of a moving source is much smaller than the sound velocity of the oceanic waveguide. In addition, a criteria for minimizing the effect of Doppler on MFP is discussed.

This paper addresses the problem of identifying the class of all stable system transfer functions that interpolate the given partial impulse response sequence. In this context, classical Pade approximations that are also stable, are shown to be a special case of this general formulation. The theory developed in this connection is utilized to obtain a new criterion for determining the model order and system parameters for rational systems, and, further, to generate nonminimum phase optimal stable rational approximatinos of nonrational systems from its impulse response sequence.

When the input-output record is available, the identification of a bilinear system is considered. It is assumed that the input is noise free and the output is contaminated by an additive noise. It is further assumed that the covariance matrix of the noise is known up to a factor of proportionality. The extended generalized total least squares (e-GTLS) method is proposed as one of the consistent estimators of the bilinear system parameters. Considering that the input is noise-free and that bilinear system equation is linear with respect to the system parameters, we extend the GTLS problem. The extended GTLS problem is reduced to an unconstrained minimization problem, and is solved by the Newton-Raphson method. We compare the GTLS method and the e-GTLS method in the point of the accuracy of the estimated system parameters.

This paper presents an algorithm for multipath channel parameter estimation by an improved Prony method. This algorithm applies a modified regularized spectral estimation to the conventional SVD Prony method. This method requires no a priori information on the number of multipath. The performance of the proposed algorithm is almost the same as that of the SVD based multipath channel parameter estimation algorithm.

We proposed to use shear waves instead of longitudinal waves in a STAM (scanning tomographic acoustic microscope system) in which the specimens are solid. For any specimen with a shear modulus, mode conversion will take place at the water-solid interface. Some of the energy of the insonifying longitudinal waves in the water will convert to shear wave energy within the specimen. The shear wave energy is detectable and can be used for tomographic reconstruction. The resolution limitation of STAM depends on the available angular view and the acoustic wavelength. While wave transmission in most solid specimens is limited to about 20°for longitudinal waves, we show that it is about twice that high for shear waves. Since the wavelength of the shear wave is shorter than that of the longitudinal wave, we are able to achieve the high resolution. In order to compare the operation of a shear-wave STAM with that of the conventional longitudinal-wave STAM we have simulated tomographic reconstruction for each. Our simulation results with aluminum specimen and back-and-forth propagation algorithm showed resolution of a shear-wave STAM is better than that of a longitudinal-wave STAM.

The goal of telecommunication may be to enable the participants in distant places to communicate with each other in an environment as if they were in the same room. This paper introduces the reason why realistic audio display is useful in telecommunication, reviews some approaches to its implementation, and proposes an audio teleconference model which realizes a two-way telecommunication with realistic sensations using binaural and auralization techniques.

In this paper, we propose a novel adaptive digital filter for tonal noise cancellation, with a frequency tracking capability. The proposed algorithm not only estimates the magnitude and phase of the tonal disturbance but also tracks its frequency, which changes in quasi-static manner. The algorithm uses the steepest descent method and the instantaneous frequency approach for the phase/magnitude estimation and frequency tracking, respectively. A number of computer simulations have been carried out in order to demonstrate the feasibility of the proposed ANC algorithm under various conditions.

This paper deals with the application of the LMS algorithm as a decision-directed adaptive equalizer in a communication receiver which also employs a sophisticated decoding scheme such as the Viterbi algorithm, in which the desired signal, hence the error, is not available until several symbol intervals later because of decoding delay. In such applications the implemented weight updating algorithm becomes DLMS and major penalty is reduced convergence speed. Therefore, every effort should by made to keep the delay as small as possible if it is not avoidable. In this paper we present a modified implementation in which the effects of the decoding delay can be avioded and perform some computer simulations to check the validity and the performance of the new implementation.