• 한국음향학회지
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• 제10권4호
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• pp.5-11
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• 1991

본 연구에서는 음성신호의 음소 단위 구분화 방법을 제안한다. 제안된 구분화 시스템은 화자 독립적이고, 음성신호에 대한 사전 정보 없이도 음소 단위로 구분화를 수행할 수 있는 특징을 갖는다. 구분화 처리는 입력 음성신호를 먼저 순수 유성을 구간과 순수 유성음이 아닌 구간으로 분리 시킨 후, 각각의 구간에 대해 세분화된 음소 단위로 분리시키는 2단계 구분화 알고리즘을 적용하였고, 이때 사용된 파라미터는 유성을 검출 파라미터, 영차 LPC 캡스트럼 계수의 시간변호 파라미터, ZCR 파라미터이다. 본 연구에서 제안한 구분화 알고리즘의 유용성을 입증하기 위해 사용한 대상어는 고립단어와 연속음성으로 구성된 어휘로서 전체 어휘중에 포함된 507개 음소에 대한 구분화율은 91.7% 이다.

• 산업경영시스템학회지
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• 제44권3호
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• pp.86-97
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• 2021

As mechatronic systems have various, complex functions and require high performance, automatic fault detection is necessary for secure operation in manufacturing processes. For conducting automatic and real-time fault detection in modern mechatronic systems, multiple sensor signals are collected by internet of things technologies. Since traditional statistical control charts or machine learning approaches show significant results with unified and solid density models under normal operating states but they have limitations with scattered signal models under normal states, many pattern extraction and matching approaches have been paid attention. Signal discretization-based pattern extraction methods are one of popular signal analyses, which reduce the size of the given datasets as much as possible as well as highlight significant and inherent signal behaviors. Since general pattern extraction methods are usually conducted with a fixed size of time segmentation, they can easily cut off significant behaviors, and consequently the performance of the extracted fault patterns will be reduced. In this regard, adjustable time segmentation is proposed to extract much meaningful fault patterns in multiple sensor signals. By considering inflection points of signals, we determine the optimal cut-points of time segments in each sensor signal. In addition, to clarify the inflection points, we apply Savitzky-golay filter to the original datasets. To validate and verify the performance of the proposed segmentation, the dataset collected from an aircraft engine (provided by NASA prognostics center) is used to fault pattern extraction. As a result, the proposed adjustable time segmentation shows better performance in fault pattern extraction.

• The Journal of the Acoustical Society of Korea
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• 제24권4E호
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• pp.128-131
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• 2005

A new algorithm for speech signal segmentation is proposed. This algorithm is based on finding successive similar frames belonging to a segment and represents it by an average spectrum. The speech signal is a slowly time varying signal in the sense that, when examined over a sufficiently short period of time (between 10 and 100 ms), its characteristics are fairly stationary. Generally this approach is based on finding these fairly stationary periods. Advantages of the. algorithm are accurate border decision of segments and simple computation. The automatic segmentations using frame averaging show as much as $82.20\%$ coincided with manually verified segmentation of CMU ARCTIC corpus within time range 16 ms. More than $90\%$ segment boundaries are coincided within a range of 32 ms. Also it can be combined with many types of automatic segmentations (HMM based, acoustic cues or feature based etc.).

• The Journal of the Acoustical Society of Korea
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• 제28권2E호
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• pp.72-78
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• 2009

Heart sound segmentation into its components, S1, systole, S2 and diastole is the first step of analysis and the most important part in the automatic diagnosis of heart sounds. Conventionally, the Shannon energy envelope peak detection method has been popularly used due to its superior performance in locating S1 and S2. Recently, the HMM has been shown to be quite suitable in modeling the heart sound signal and its use in segmenting the heart sound signal has been suggested with some success. In this paper, we compared the two methods for heart sound segmentation using a common database. Experimental tests carried out on the 4 different types of heart sound signals showed that the segmentation accuracy relative to the manual segmentation was 97.4% in the HMM based method which was larger than 91.5% in the peak detection method.

• 대한인간공학회지
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• 제29권4호
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• pp.667-671
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• 2010

Runs test is a mathematical tool to test the stationarity of electromyographic (EMG) signals. The purpose of this study is to investigate the effects of segmentation size on the stationarity of EMG signals in runs test. Six subjects participated in this experiment and performed isometric trunk exertions for twenty seconds at the load level of 25% and 50% MVC. The signals extracted from the erector spinae muscles were divided into the intervals of 1000ms and the stationarity of the signal in each interval was tested by the runs test. In this test, seven segmentation sizes such as 1.0, 2.0, 3.9, 7.8, 15.6, 31.3 and 62.5ms were applied. Additionally, two stationarity tests of reverse arrangements test and modified reverse arrangements test were used to verify the results of the runs test. In results, the segmentation size of 62.5ms showed the similar results with the other stationarity tests. However, the stationarity values among there tests were different each other when segmentation sizes other than 62.5ms were used. These results indicated the effect of segmentation size in runs test that needs to be considered to have consistent and sensitive result in stationarity test.

• 음성과학
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• 제12권1호
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• pp.135-142
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• 2005

An important aspect of video indexing is the ability to segment video into meaningful segments, i.e., content-based video segmentation. Since the audio signal in the sound track is synchronized with image sequences in the video program, a speech signal in the sound track can be used to segment video into meaningful segments. In this paper, we propose a new approach to content-based video segmentation. This approach uses closed caption to construct a recognition network for speech recognition. Accurate time information for video segmentation is then obtained from the speech recognition process. For the video segmentation experiment for TV news programs, we made 56 video summaries successfully from 57 TV news stories. It demonstrates that the proposed scheme is very promising for content-based video segmentation.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
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• pp.306-308
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• 2005

The vehicle detection method using pulse radar has the advantage of maintenance in comparison with loop detection method. We have the information about the vehicle being and position by dividing the signals into sectors in accordance with SSC method, and by applying the discriminant function based on stochastical data. We also reduce the signal processing time.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2216-2219
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• 2003

Accurate speech region detection and automatic syllable segmentation is important part of speech recognition system. In automatic speech recognition system, they are needed for the purpose of accurate recognition and less computational complexity, In this paper, we Propose improved syllable segmentation method using ramp edge detection method and residual signal Peak energy. These methods were used to ensure accuracy and robustness for endpoint detection and syllable segmentation system. They have almost invariant response to various background noise levels. As experimental results, we obtained the rate of 90.7％ accuracy in syllable segmentation in a condition of accurate endpoint detection environments.

• 융합신호처리학회논문지
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• 제10권1호
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• pp.32-39
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• 2009

동화상의 부호화 효율향상을 위하여 유전적 알고리즘(Genetic Algorithm: GA)을 이용하여 영역 분할하는 방법을 제안한다. 유전적 알고리즘은 함수치만을 이용하여 큰 탐색공간으로부터 최적의 조합을 축차적으로 찾아내는 방법이다. 이동추정과 영역분할을 동시에 진행함으로써, 이동 벡터를 화면내의 작은 블록이나 화소의 각각에 할당하고, 그것을 부호화 정보량과 신호 대 잡음비의 관계로부터 최적화 문제로 변환할 수 있다. 즉, 이동보상예측 부호화에는 영역분할과 이동 추정은 서로 밀접하게 관계되어 있다. 이것은 부호량과 S/N비를 최적화하는 것으로서 화면 속의 각 블록에서 이동 벡타를 최적의 상태로 배치하는 것이다. 그러므로, 본 논문에서는 최적인 영역분할 결과를 얻기 위하여 GA의 데이터형과 그 데이터의 처리 방법에 대해서 검토하였다. 또한, 테스트 화상을 이용한 컴퓨터시뮬레이션을 통하여 제안 방법의 유효성을 확인하였다.

• 한국음향학회지
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• 제19권4호
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• pp.58-68
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• 2000

본 논문에서는 폴리포닉 오디오 신호에 대한 정현파 모델링 방법을 제안한다. 정현화 모델링을 폴리포닉 오디오 신호에 적용하는데 있어서 가장 큰 문제점은 스펙트럼 분석을 위한 분석 윈도우의 크기를 결정할 수 없다는 것이다. 또한 고음질의 합성음을 위해서는 악기음의 특성을 결정짓는 어택이 잘 보존되어야 한다. 본 논문에서는 입력 신호를 6개의 옥타브 벤드 구조의 다중 해상도 필터 뱅크를 통과시키고, 각 서브벤드 신호에 대해 서로 다른 크기의 분석 윈도우를 적용시킴으로써 폴리포닉 오디오 신호에 대한 분석 윈도우 크기 결정 문제를 해결한다. 정현파 모델링에서 발생하는 어택과 같은 천이 구간에서의 퍼짐 현상을 개선하기 위해 각 서브밴드 신호에 동적 세그맨테이션 방법을 적용하여 천이 구간 근처에서는 분석과 합성 프레임 크기를 작게 하는 방법을 사용한다. 이 방법을 통해 서브밴드 신호의 구간별 시간-주파수 특성에 따라 적절한 크기의 윈도우를 선택할 수 있다. 동적 세그멘테이션 방법으로는 기존의 방법보다 계산량과 성능 면에서 더 나은 특성을 보이는 방법을 제안한다. 여러가지 폴리포닉 오디오 신호에 대한 시뮬레이션 결과 제안한 정현파 모델링 방법이 음질의 손상 없이 원래 신호를 잘 복원할 수 있음을 확인하였다.