• Journal of the Korean Society for Nondestructive Testing
• /
• v.26 no.5
• /
• pp.343-352
• /
• 2006

In the present paper, the nondestructive test method was suggest to establish the bonding status of a motor case assembly composed of a steel motor case, adiabatic rubber layer and an ablative composite tube with strain data, AE(acoustic emission) signals and UT(ultrasonic test) data. And, finite element analysis was conducted to verify quantitatively the bonding status of motor case assembly under inner pressure loading. The bonding status could be judged whether the bonding status is perfect or contact condition by the data correlation study with AE signals and strain data measured from air pressure test. And, to classify the bonding status of motor case and rubber layer among bonding layers, UT method was also applied. From this study, the bonding status could be classified and detected into fourth types for all bonding layers as follows: (1) initial un-bonding, (2) perfect do-bonding during an air pressure test, (3) partially de-bonding during an air pressure test, and (4) perfect bonding.

• Journal of the Korean Society for Nondestructive Testing
• /
• v.26 no.4
• /
• pp.220-230
• /
• 2006

Fracture behavior of a monolithic PZT and a plate-type piezoelectric composite actuator (PCA) has been investigated under a bending load at three points by an acoustic emission (AE) monitoring. AE signal from a monolithic PZT at the maximum bending load shows the characteristics of high amplitude and long duration with a low frequency band of $100{\sim}230kHz$ which is confirmed by fast Fourier transform (FFT). For a PCA, it is concluded that AE signals with high amplitude over 80dB and low dominant frequency band of $170{\sim}223kHz$ emitted in the stage I are due to the brittle fracture in the PZT layer and the delamination between the PZT layer and the adjacent fiber composite layer. Based on the above analysis of AE behavior and damage observations with an optical microscopy and a scanning electron microscopy, AE characteristics related to fracture behavior of asymmetrically laminated PCA have been elucidated.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2014.06a
• /
• pp.5-8
• /
• 2014

본 논문에서는 비음수 행렬 분해 (NMF) 기법을 이용하여 단일 채널에서 배경음 성분을 추출하는 알고리즘에 대해 서술한다. 이러한 배경음 성분 추출은 오디오 업믹싱 시스템을 고려하여 개발되었으며, 기존의 연구를 통하여 분리된 배경음 신호가 서라운드 채널 혹은 상방향 채널에 적용될 경우 청취자의 공간감을 향상시킬 수 있다는 사실이 이미 확인된 바 있다. 다만 기존의 기법은 음향 신호를 모두 축적하여 일괄적으로 처리해야 한다는 단점이 있어, 스트리밍 시스템이나 디지털 신호 프로세서 등을 이용한 시스템에서 사용될 수 없는 단점이 있다. 본 논문에서는 이를 해소하기 위하여 실시간 비음수 행렬 분해 기법을 이용한 배경음 추출 시스템을 고안하여 실험하였다. 실험 결과 실시간 배경음 추출 기법이 신호의 후반부에서는 원하는 대로 동작하나, 초중반에 기저가 과도하게 설정되는 문제점이 있음을 확인할 수 있었으며, 이에 대한 해결이 향후 연구 과제가 될 것이다.

• The Journal of the Acoustical Society of Korea
• /
• v.13 no.6
• /
• pp.91-97
• /
• 1994

In speech processing, the waveform codings are concerned with simply preserving the waveform of signal through a redundancy reduction process. In the case of speech synthesis, the waveform codings with high quality are mainly used to the synthesis by analysis. Because the parameters of this coding are not classified as both excitation and vocal tract, it is difficult to apply the waveform coding to the synthesis by rule. Thus, in order to apply the waveform coding to synthesis by rule, it is necessary to alter the pitches. In this paper, we proposed a new pitch alteration method that can change the pitch period in waveform coding by dividing the speech signals into the vocal tract and excitation parameters. This method is a time-frequency domain method preserving the phase component of the waveform in time domain and the magnitude component in frequency domain. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing. In case of using the algorithm, we can obtain spectrum distortion with $2.94\%$. That is, the spectrum distortion is decreased more $5.06\%$ than that of the pitch alteration method in time domain.

• The Journal of the Acoustical Society of Korea
• /
• v.38 no.4
• /
• pp.427-432
• /
• 2019

In this paper, we propose a home monitoring system using sound event detection based on a bidirectional gated recurrent neural network for the hard-of-hearing. First, in the proposed system, packet loss concealment is used to recover a lost signal captured through wireless sensor networks, and reliable channels are selected using multi-channel cross correlation coefficient for effective sound event detection. The detected sound event is converted into the text and haptic signal through a harmonic/percussive sound source separation method to be provided to hearing impaired people. Experimental results show that the performance of the proposed sound event detection method is superior to the conventional methods and the sound can be expressed into detailed haptic signal using the source separation.

• The Journal of the Acoustical Society of Korea
• /
• v.42 no.3
• /
• pp.203-213
• /
• 2023

In medical ultrasound imaging systems, Side lobes may appear if signals outside the imaging point are not completely removed during receive focusing. If the time signal of the side lobe overlaps with the time signal (main lobe) from the image point, it is difficult to completely remove it using filter processing in the time domain. However, In the receive focusing process, when time-channel signals are Fourier-transformed, the main lobe and side lobe signals are spatially separated in the spectral domain. Therefore, the side lobes can be suppressed by multiplying the image with magnitude weights, which are determined by the magnitudes of the main and side lobes calculated in the spectral domain. In addition, when the main lobe and the side lobe spectrum are adjacent, the distance weight was applied based on the distance between them. In a 5 MHz ultrasound imaging system using a 64-channel linear transducer, point reflector and speckle images with cysts of various brightness were synthesized and weights were applied to the ultrasound image. Using computer simulations, we confirmed that the side lobes were greatly reduced without affecting the spatial resolution in the point reflector image, and the contrast was significantly improved in the cyst image with computer simulations.

• The Journal of the Acoustical Society of Korea
• /
• v.8 no.4
• /
• pp.53-59
• /
• 1989

A new proposed method can separate heart sounds and lung sounds by the realization of adaptive noise canceler using adaptive lattice Wiener filter in contrast to adaptive transversal LMS filter and high pass filter as before. Lung sounds and ECG signal are detected for this purpose, and especially the second heart sounds are reduced by finding T wave location with a T wave seeking algorithm. As a result, for heart sounds reduction It was found that adaptive transversal LMS filter required 100-200's orders, 75-100's orders In adaptive transversal MLMS filter, and only 10-20's orders in adaptive lattice Wiener filter. Adaptive filtering technique has shown greater accuracy than high pass filtering without loss of low frequency component.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.11 no.3
• /
• pp.978-983
• /
• 2010

Acoustic target classification in wireless sensor network is important research at environmental surveillance, invasion surveillance, multiple target separation. General sensor node signal processing methods concentrated on received signal energy based target detection and received raw signal compression. The former is not suited to target classification because of almost every target information are lost except target energy. The latter bring down life-time of sensor node owing to high computational complexity and transmission energy. In this paper, we introduce an feature extraction algorithm for acoustic target classification in wireless sensor network which has time and frequency information. The proposed method extracts time information and de-noised target classification information using wavelet decomposition step. This method reduces communication energy by 28% of original signal and computational complexity.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.1
• /
• pp.49-54
• /
• 2016

In this paper, we propose a vocal separation method using weighted ${\beta}$-order minimum mean wquare error estimation (WbE) based on kernel back-fitting algorithm. In spoken speech enhancement, it is well-known that the WbE outperforms the existing Bayesian estimators such as the minimum mean square error (MMSE) of the short-time spectral amplitude (STSA) and the MMSE of the logarithm of the STSA (LSA), in terms of both objective and subjective measures. In the proposed method, WbE is applied to a basic iterative kernel back-fitting algorithm for improving the vocal separation performance from monaural music signal. The experimental results show that the proposed method achieves better separation performance than other existing methods.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.5
• /
• pp.353-360
• /
• 2017

The single sensor based acoustic reflection control system separates the incident and reflected signals from the single sensor output, and reduces the reflected signal by generating an out-of-phase signal from the incident signal component. In this paper, we propose an optimal filter design method for a single sensor based reflection control system. In the proposed method, it is shown that the optimum control filter design is possible by using the measured impulse responses of the reflection and control paths. The reflection control algorithm based on the proposed optimal filter achieves better performance than the conventional adaptive filter-based algorithm and effectively controls the reflection without the initial convergence time. We performed computer simulations using the signals obtained in a 1-dimensional acoustic duct environment, and from the simulation results, it was confirmed that the proposed optimal filter has robust performance even in noisy environment.