• Journal of Advanced Marine Engineering and Technology
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• 제36권2호
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• pp.258-266
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• 2012

음향방출(Acoustic Emission, AE) 시스템은 최근 조기 결함 검출 시스템 개발을 위해 적용되고 있으며, 그에 따르는 신호처리 기법에 대한 문제를 해결하기 위해 많은 노력을 기울이고 있다. 신호처리 기법 중 포락처리(Envelope analysis)가 베어링 결함 분석에 사용되고, Wavelet Transform은 기어 등의 결함 분석에 용이한 것으로 알려져 있다. 하지만 여전히 AE 신호를 위한 신호처리 기법은 불확실하다. 따라서 본 논문에서는 AE 시스템을 적용한 조기 결함 검출 시스템 개발을 위한 사전 연구로, AE 신호를 분석하기 위한 신호처리 기법으로 Hilbert Transform(HT)과 Hilbert-Huang Transform(HHT)에 대해 비교 분석한다. AE 신호는 피로시험을 통해 취득되었으며, 취득된 AE 신호를 두 신호처리 기법을 적용하여 주파수 및 시간 신호에 대해 분석하였다. HT에 비해 HHT가 시간-주파수 영역에 대해 결과를 나타내기 때문에 좀 더 명확한 특징을 보이는 데에 반해 신호처리 시간 및 필터링에 대한 단점을 보이고 있음을 확인하였다.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2005년도 추계종합학술대회
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• pp.451-454
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• 2005

현재, 신호를 해석하기 위한 많은 방법들이 제시되고 있으며, 대표적인 방법으로는 퓨리에 변환과 웨이브렛 변환이 있다. 이러한 방법들에서, 퓨리에 변환은 모든 주파수 범위에 대해 cosine과 sine 파형의 조합으로써 신호를 표현하지만, 신호 내에서 특정 주파수 성분이 발생한 시간정보를 제공하지 않으며, 분석 신호의 전체적인 특징만을 나타낸다. 따라서 이러한 한계를 극복하기 위해, 다중해상도 해석이 가능한 웨이브렛 변환이 음성과 영상처리, 컴퓨터 비전 등의 광범위한 분야에서 응용되고 있다. 그리고 웨이브렛 변환은 스케일 변수에 따라 변화하는 윈도우를 사용하여 시간-주파수 국부성을 나타낸다. 본 논문에서는 cosine과 sine 형태의 웨이브렛을 사용하여, 퓨리에 변환의 새로운 접근법을 제시하였으며, 주파수-시간 평면의 유한한 지점에서 신호의 특징을 분석하였다.

• Earthquakes and Structures
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• 제9권1호
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• pp.173-194
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• 2015

Site response analysis is an important topic in earthquake engineering. A time-domain numerical method called as one-dimensional (1D) finite element artificial boundary method is proposed to simulate the homogeneous plane elastic wave propagation in a layered half space subjected to the obliquely incident plane body wave. In this method, an exact artificial boundary condition combining the absorbing boundary condition with the inputting boundary condition is developed to model the wave absorption and input effects of the truncated half space under layer system. The spatially two-dimensional (2D) problem consisting of the layer system with the artificial boundary condition is transformed equivalently into a 1D one along the vertical direction according to Snell's law. The resulting 1D problem is solved by the finite element method with a new explicit time integration algorithm. The 1D finite element artificial boundary method is verified by analyzing two engineering sites in time domain and by comparing with the frequency-domain transfer matrix method with fast Fourier transform.

• 한국정보통신학회논문지
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• 제16권7호
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• pp.1437-1446
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• 2012

본 논문에서는 스테레오 카메라 환경에서 고속으로 깊이 정보를 얻기 위한 응용분야에 적합한 스테레오 정합 기법을 제안하고자 한다. 이러한 조건을 만족하기 위해서 DWT 영역에서의 주파수 정보와 가변 정합창을 이용하는 적응적인 스테레오 정합 기법을 제안한다. 공간 영역에서 영상의 국부적인 특성을 분석하여 정합창의 크기를 결정하고, 주파수 영역에서 영상의 주파수 특성을 분석하여 정합창의 형태 및 스케일링 요소를 결정한다. 주파수 영역에 대한 정보를 이용하기 위해서 로컬 DWT와 전역 DWT를 활용하는 기법을 모두 적용하였다. 본 논문은 스테레오 정합을 위한 제안한 기법은 유사한 수준의 복잡도에서 고정 정합창 기반의 기법과 비교할 때 PSNR이 향상되는 것을 확인하였다.

• IEIE Transactions on Smart Processing and Computing
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• 제4권1호
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• pp.22-27
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• 2015

Transform-domain JND (Just Noticeable Difference)-based for PVC (Perceptual Video Coding) is often performed in quantization processes to effectively remove perceptual redundancy. This study examined the JND-suppression effects on quantized coefficients of transform in HEVC (High Efficiency Video Coding). To reveal the JND-suppression effect in quantization, the properties of the floor functions were used for modeling the quantized coefficients, and a JND-adjustment process in an HEVC-compliant PVC scheme was used to tune the JND values by analyzing the JND suppression effect. In the experimental results, the bitrate reduction decreases slightly, but the PSNR and perceptual quality are improved significantly when the proposed JND adjustment process is applied.

• 음성과학
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• 제7권1호
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• pp.17-29
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• 2000

The STC(Sinusoidal Transform Coding) is a vocoding technique that uses a sinusoidal speech model to obtain high- quality speech at low data rate. It models and synthesizes the speech signal with fundamental frequency and its harmonic elements in frequency domain. To reduce the data rate, it is necessary to represent the sinusoidal amplitudes and phases with as small number of peaks as possible while maintaining the speech quality. As a basic research to develop a low-rate speech coding algorithm using the sinusoidal model, in this paper, we investigate the speech quality depending on the number of sinusoids. By varying the number of spectral peaks from 5 to 40 speech signals are reconstructed, and then their qualities are evaluated using spectral envelope distortion measure and MOS(Mean Opinion Score). Two approaches are used to obtain the spectral peaks: one is a conventional STFT (Short-Time Fourier Transform), and the other is a multiresolutional analysis method.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 ITC-CSCC -3
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• pp.2035-2038
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• 2002

The necessity of the filter coefficients compensation for the LTJ adaptive filter was explained generally and easily by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%

• Journal of Information Processing Systems
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• 제1권1호
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• pp.32-35
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• 2005

Image denoising is basic work for image processing, analysis and computer vision. This paper proposes a novel algorithm based on wavelet threshold for image denoising, which is combined with the linear CLS (Constrained Least Squares) filtering and thresholding methods in the transform domain. We demonstrated through simulations with images contaminated by white Gaussian noise that our scheme exhibits better performance in both PSNR (Peak Signal-to-Noise Ratio) and visual effect.

• ETRI Journal
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• 제31권1호
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• pp.1-9
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• 2009

Frequency domain analysis is a fundamental procedure for understanding the characteristics of visual data. Several studies have been conducted with 2D videos, but analysis of stereoscopic 3D videos is rarely carried out. In this paper, we derive the Fourier transform of a simplified 3D video signal and analyze how a 3D video is influenced by disparity and motion in terms of temporal aliasing. It is already known that object motion affects temporal frequency characteristics of a time-varying image sequence. In our analysis, we show that a 3D video is influenced not only by motion but also by disparity. Based on this conclusion, we present a temporal anti-aliasing filter for a 3D video. Since the human process of depth perception mainly determines the quality of a reproduced 3D image, 2D image processing techniques are not directly applicable to 3D images. The analysis presented in this paper will be useful for reducing undesirable visual artifacts in 3D video as well as for assisting the development of relevant technologies.

• 대한음성학회지:말소리
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• 제61호
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• pp.63-74
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• 2007

Discrimination of music and speech from the multimedia signal is an important task in audio coding and broadcast monitoring systems. This paper deals with the problem of feature parameter extraction for discrimination of music and speech. The wavelet transform is a multi-resolution analysis method that is useful for analysis of temporal and spectral properties of non-stationary signals such as speech and audio signals. We propose new feature parameters extracted from the wavelet transformed signal for discrimination of music and speech. First, wavelet coefficients are obtained on the frame-by-frame basis. The analysis frame size is set to 20 ms. A parameter $E_{sum}$ is then defined by adding the difference of magnitude between adjacent wavelet coefficients in each scale. The maximum and minimum values of $E_{sum}$ for period of 2 seconds, which corresponds to the discrimination duration, are used as feature parameters for discrimination of music and speech. To evaluate the performance of the proposed feature parameters for music and speech discrimination, the accuracy of music and speech discrimination is measured for various types of music and speech signals. In the experiment every 2-second data is discriminated as music or speech, and about 93% of music and speech segments have been successfully detected.