• 비파괴검사학회지
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• 제20권6호
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• pp.501-510
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• 2000

초음파 에코파형의 시간-주파수해석법으로 Wigner 분포와 웨이브렛 변환 등과 같은 새로운 신호처리 기법이 비파괴평가 분야에 널리 응용되고 있다. 본 연구에서는 웨이브렛 변환(wavelet transform)에 의한 음속과 감쇠계수의 주파수의존성과 초음파 결함신호의 잡음제거의 유용성 유무를 실험적으로 검증하였다. Gabor 함수를 웨이브렛의 기본함수로 사용하였다. 초음파의 에코파형에 포함된 각 주파수성분의 속도와 감쇠계수의 주파수의존성을 추정할 수 있었으며, 초음파탐상에서 결함의 검출능 향상과 결함크기 산정의 정량화에 접근하기 위해 웨이브렛 변환에 의한 S/N비 신호처리 시뮬레이션 결과를 오스테나이트강 스테인레스 용접부에 가공한 EDM 노치의 초음파 결함신호에 적용한 결과 임상에코를 저감하고 S/N비를 개선하는 것이 가능하였다.

• ETRI Journal
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• 제35권5호
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• pp.748-756
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• 2013

This paper addresses the problem of online hop timing detection and frequency estimation of multiple frequency-hopping (FH) signals with antenna arrays. The problem is deemed as a dynamic one, as no information about the hop timing, pattern, or rate is known in advance, and the hop rate may change during the observation time. The technique of particle filtering is introduced to solve this dynamic problem, and real-time frequency and direction of arrival estimates of the FH signals can be obtained directly, while the hop timing is detected online according to the temporal autoregressive moving average process. The problem of network sorting is also addressed in this paper. Numerical examples are carried out to show the performance of the proposed method.

• 한국음향학회지
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• 제40권5호
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• pp.401-407
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• 2021

연속 송수신 소나 기술은 표적 탐지 시간간격이 짧아 표적을 탐지하고 추적하는 데 효과적인 기술이다. 연속 송수신 소나 시스템에 적합한 Generalized Sinusoidal Frequency Modulation(GSFM)펄스는 펄스 간 직교성을 유지하면서 높은 시간-주파수 곱을 얻을 수 있는 것으로 알려져 있다. 그러나 가속도를 갖는 표적이 존재할 경우 수신된 GSFM 펄스를 상관처리하는 적정한 시간 길이를 산출하는 방법에 대해서는 알려진 바가 없다. 본 논문에서는 GSFM 펄스를 사용하는 연속 송수신 소나 시스템에서 수신신호를 정합처리할 때 적합한 시간 길이를 상관도에 근거하여 산출하는 방법을 제안한다. 제안된 방법은 표적의 가속도에 따른 상관도를 계산하고 상관도에 따른 신호처리 길이를 산출한다. 부엽의 준위를 통해 제안된 방법으로 산출된 신호처리 길이를 적용할 경우 안정적인 탐지 성능을 얻을 수 있음을 보인다.

• 제어로봇시스템학회논문지
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• 제4권4호
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• pp.517-524
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• 1998

This paper discusses the development of a reactive power controller using digital signal processing. Digital Signal Processing is the technique of using digital devices to Process continuous signals or data, often in real-time. And DSP algorithms are associated with a discrete time interval between input samples. When one designs a digital filter, one can use a Laplace transform to determine the continuous time frequency response. The corresponding discrete time transform is called Z transform and depends upon discrete samples of the input spaced equally in time. The objectives of this paper are to minimize real power losses and improve the power factor of a given system. Also, the implementation of a direct-form non recursive filter on the TMS320C31 has been described. The application of this microprocessor-based controller using DSP on test system reveals its numerous advantages. Performance and features of the controller for the reactive power control are analyzed.

• Structural Engineering and Mechanics
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• 제66권2호
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• pp.185-193
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• 2018

In this study, low- and high-frequency structure behaviors were identified and a systematic analysis procedure was proposed using noisy GPS data from a 165-m-high tower in ${\dot{I}}stanbul$, Turkey. The raw GPS data contained long- and short-periodic position changes and noisy signals at different frequencies. To extract the significant results from this complex dataset, the general structure and components of the GPS signal were modeled and analyzed in the time and frequency domains. Uncontrolled jumps and deviations involving the signal in the time domain were pre-filtered. Then, the signal was converted to the frequency domain after applying low- and high-pass filters, and the frequency and periodic component values were calculated. The spectrum of the tower motion obtained from the filtered GPS data had dominant peaks at a low frequency of $1.15572{\times}10-4Hz$ and a high frequency of 0.16624 Hz, consistent with two equivalent GPS datasets. Then, the signal was reconstructed using inverse Fourier transform with the dominant low frequency values to obtain filtered and interpretable clean signals. With the proposed sequence, processing of noisy data collected from the GPS receivers mounted very close to the structure is effective in revealing the basic behaviors and features of buildings.

• Journal of Information Processing Systems
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• 제17권5호
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• pp.1004-1019
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• 2021

To solve the problems of the low image contrast, fuzzy edge details and edge details missing in noisy image fusion, this study proposes a noisy infrared and visible light image fusion algorithm based on non-subsample contourlet transform (NSCT) and an improved bilateral filter, which uses NSCT to decompose an image into a low-frequency component and high-frequency component. High-frequency noise and edge information are mainly distributed in the high-frequency component, and the improved bilateral filtering method is used to process the high-frequency component of two images, filtering the noise of the images and calculating the image detail of the infrared image's high-frequency component. It can extract the edge details of the infrared image and visible image as much as possible by superimposing the high-frequency component of infrared image and visible image. At the same time, edge information is enhanced and the visual effect is clearer. For the fusion rule of low-frequency coefficient, the local area standard variance coefficient method is adopted. At last, we decompose the high- and low-frequency coefficient to obtain the fusion image according to the inverse transformation of NSCT. The fusion results show that the edge, contour, texture and other details are maintained and enhanced while the noise is filtered, and the fusion image with a clear edge is obtained. The algorithm could better filter noise and obtain clear fused images in noisy infrared and visible light image fusion.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2003년도 춘계학술대회논문집
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• pp.1009-1009
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• 2003

VIPER a new tool for the VIsual PERception of sound quality and for sound design will be presented. Requirement for the visualization of sound quality is a signal analysis modeling the information processing of the ear. The first step of the signal processing implemented in VIPER, calculates an auditory spectrogram by a filter bank adapted to the time- and frequency resolution of the human ear. The second step removes redundant information by extracting time- and frequency contours from the auditory spectrogram in analogy to contours of the visual system. In a third step contours and/or auditory spectrogram can be resynthesised confirming that only aurally relevant information were extracted. The visualization of the contours in VIPER allows intuitively to grasp the important components of a signal. Contributions of parts of a signal to the overall quality can be easily auralized by editing and resynthesising the contours or the underlying auditory spectrogram. Resynthesis of time contours alone allows e.g. to auralize impulsive components separately from the tonal components. Further processing of the contours determines tonal parts in form of tracks. Audible differences between two versions of a sound can be visually inspected in VIPER through the help of auditory distance spectrograms. Applications are shown for the sound design of several interior noises of cars.

• 한국컴퓨터정보학회논문지
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• 제11권4호
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• pp.55-60
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• 2006

문자 영역 추출을 위해서 FFT와 신경망을 이용한 방법을 본 논문에서 제안하고자 한다. 일반적으로 문자 영역은 고주파 영역에서 발견되므로 FFT를 이용하여 이 특징을 추출할 수 있다. 문자(고 주파) 영역과 비 문자(저 주파) 영역을 신경망에 학습을 시킨다. 신경망에 고주파 영역을 입력으로써 후보 영역을 추출한다. 그리고 최종 문자 영역은 후보 영역 검증을 통하여 추출된다. 실험 결과 후보 영역 추출은 학습된 경우 100% 추출율을 보여주고 있으며, 검증을 통한 후보 영역 추출율은 95%임을 알 수 있었다. 제안된 알고리즘의 장점은 알고리즘의 단순성과 실시간 처리에 있다.

• 융합신호처리학회논문지
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• 제1권1호
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• pp.7-14
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• 2000

최근 들어 초고속 정보통신기기 소자와 디지털 신호처리 기술에 힘입어 적응잡음제거 기술이 비약적인 발전을 이룩하게 되었다. 그러나 실제로 적응잡음제거, 에코제거 등의 여러 응용에서 만족할 만한 결과를 얻기 위해서는 수천이 넘는 적응필터계수가 요구된다. 계수의 수가 많은 적응 FIR 필터의 복잡한 계산 량을 줄이고 빠른 수렴을 얻기 위한 시도로 본 연구에서는 주파수영역 적응필터와 다중지연 주파수영역 적응필터 알고리즘을 실시간 처리 관점에서 서로 비교하였다. 긴 탭의 주파수영역 적응필터에서 발생하는 시간지연은 다중지연 주파수영역 적응필터를 적용하여 개선하려다. 또한 일반 적응잡음제거환경과 Cross talk 환경에 주파수영역 적응필터을 적용하여 그 알고리즘의 실시간 처리 가능성을 dSPACE 1103 보드를 통하여 확인하였다

• 한국안전학회지
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• 제6권2호
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• pp.21-30
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• 1991

In this paper, we study effect of waveness at the optical surface roughness measurement. In generally, waveness components cause errors in calculation of the roughness value of metal surface. We study about surface roughness signals In the frequency domain for separate two signal component of real roughness and waveness by digital signal processing methods. Thereafter, determine low and high Component of frequency spectrum. By this separating frequency value we design liner low and high pass filter which cutoff frequency is 1 Hz. After this process, converted each filtered spectrum by inverse discrete fourier transformation to time domain waveness and real roughness signals We calculate surface roughness value from filtered roughness signals. For evaluate this method, we use five specimens roughness signal which obtained from optical surface roughness measuring system in 3mm/s moving speed with 0.1 mm laser beam spot size As a result, we obtain more linerized roughness value than that of unfiltered roughness signals.