• The Journal of the Convergence on Culture Technology
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• v.8 no.5
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• pp.713-721
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• 2022

In order to use the LED video wall in the broadcasting studio, there are a few things to be aware of. First, since the pixels are closely arranged, a moire phenomenon may occur due to a short arrangement period, and second, the distance between pixels (pixel pitch) may be recorded on the image sensor of the broadcasting camera. When moire occurs or pixel pitch is observed, viewers feel uncomfortable. Moire effect can be reduced by adjusting the shooting distance or angle of the camera, but in order to prevent the pixel pitch from being recorded on the image sensor, secure a sufficient distance between the LED video wall and camera. even when the distance secured, the zoom lens used in the broadcasting studio must be operated by appropriately changing the magnification. If the focal length is changed by changing the magnification to obtain a desired angle of view, the pixel pitch may be unintentionally recorded. In this study we propose the range that the pixel pitch is not observed while changing the magnification ratio of the zoom lens when the distance from the video wall is sufficiently secured. The content was played back on the LED video wall and the LED video wall was recorded on the server using an HD camera equipped with a B4 mount zoom lens

• 한국방재학회:학술대회논문집
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• 2009.02b
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• pp.106.2-106.2
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• 2009

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.1
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• pp.23-28
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• 2006

Pitch period that is a important factor in speech signal processing is used in various applications such as speech recognition, speaker identification, speech analysis and synthesis. So many pitch detection algorithms have been studied until now. AMDF which is one of pitch period detection algorithms chooses the time interval from valley point to valley point as pitch period. In selection of valley point to detect pitch period, complexity of the algorithm is increased. So in this paper we proposed the simple algorithm using rotation transform of AMDF that detects global minimum valley point as pitch period of speech signal and compared it with existing methods through simulation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.515-519
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• 2005

Pitch period that is a important factor in speech signal processing is used in various applications such as speech recognition, speaker identification, speech analysis and synthesis. So many pitch detection algoritms have been studied until now. AMDF which is one of pitch period detection algorithms chooses the time interval from valley point to valley point as pitch period. In selection of valley point to detect pitch period, complexity of the algoritm is increased. So in this paper we proposed the simple algorithm using modified AMDF that detects global minimum valley point as pitch period of speech signal and compared existing methods with it through simulation.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.2
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• pp.127-131
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• 2009

In the digital waveguide theory, traveling waves are represented by general solution to the wave equation that is second-order linear partial differential equation. The movement of these waves can be implemented using only delay lines. An unit delay in the general digital waveguide describes a sampling time interval. However, in the space-based digital waveguide the unit delay implies the spatial sampling distance. In consideration of these differences between two models, it is known that the space-based digital waveguide model is adequate to synthesize pitch-shifted sounds such as vibrato because the propagation distance can be directly control. In this paper, the time-based digital waveguide model which also synthesizes pitch-shifted sounds is proposed and compared with space-based digital waveguide.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.1019-1022
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• 2005

As recent information communication technology is rapidly developed, a lot of researches related to speech signal processing have been processed. So pitch period is applied as important factor to many application fields such as speech recognition, speaker identification, speech analysis and synthesis. Therefore, many algorithms related to pitch detection have been proposed in time domain and frequency domain and AMDF(average magnitude difference function) which is one of pitch detection algorithms in time domain chooses time interval from valley to valley as pitch period. But, in selection of valley point to detect pitch period, complexity of the algorithm is increased. So in this paper we proposed pitch detection algorithm using rotation transform of AMDF, that taking the global minimum valley point as pitch period and established a threshold about the phoneme in beginning portion, to exclude pitch period selection. and compared existing methods with proposed method through simulation.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.257-259
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• 2012

이 논문에서는 실험을 통해서, 기존에 제안하였던 쿼리-바이-싱잉/허밍 (Query-by-singing/humming, QbSH) 시스템의 매칭 엔진의 강인성을 검증하고 그 결과를 제시한다. QbSH 시스템은 디지털 음악의 사용이 보편화되면서 음악 검색의 방법으로 많은 연구가 진행되어 오고 있다. QbSH 시스템은 입력으로부터 멜로디의 특징을 추출하는 부분과 추출된 특징을 매칭하는 부분으로 나눌 수 있는데, 매칭 단계에서 특징이 추출된 두 개의 멜로디 사이의 유사도 또는 거리를 계산하여 가장 유사한 멜로디를 데이터베이스에서 찾게 된다. 이 논문에서는 이 중, 기존에 제안하였던 매칭 엔진 부분의 강인성을 알아보기 위해서 입력으로부터 멜로디의 피치 시퀀스를 추출하는 과정을 간략히 하여 그 결과를 살펴보았다. 즉, 기존에 제안한 매칭 엔진이 특정한 피치 시퀀스 추출 과정에서만 좋은 성능을 보이는 게 아님을 실험을 통해서 살펴보았다. 실험 결과, 피치 시퀀스를 추출하는 과정이 극도로 간략해지더라도, 매칭 엔진을 좋은 성능을 보여주었다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.383-391
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• 2009

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. The waveform which is stationary region of voiced speech is very correlated by pitch period. So we can estimate the SNR by correlation of near waveform after dividing a frame for each pitch. For unvoiced speech signal, vocal track characteristic is reflected by noise, so we can estimate SNR by using spectral distance between spectrum of received signal and estimated vocal track. Lastly, energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced.

• Journal of Broadcast Engineering
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• v.25 no.4
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• pp.576-586
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• 2020

Analyzing baseball games using sensors such as radars or riders is expensive. In this paper, we propose an algorithm to detect pitch shots and hit shots using baseball video and to generate ball trajectories within hit shots using camera movement. After the pitch shot and the hit shot detection using object detection and optical flow, we generate the transformation relationship between frames and ball locations in the frame, and calculates the ball trajectory. The performance of the proposed method is evaluated for three KBO baseball video sequences, and the detection accuracy and detection rate of pitch shot and hit shot were within 89-95 [%], and the average error for shot range was 13.6[m], The direction error was 7.5° and foul classification accuracy was 98.6%.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.63-69
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• 2001

A corpus-based lip sync algorithm for synthesizing natural face animation is proposed in this paper. To get the lip parameters, some marks were attached some marks to the speaker's face, and the marks' positions were extracted with some Image processing methods. Also, the spoken utterances were labeled with HTK and prosodic information (duration, pitch and intensity) were analyzed. An audio-visual corpus was constructed by combining the speech and image information. The basic unit used in our approach is syllable unit. Based on this Audio-visual corpus, lip information represented by mark's positions was synthesized. That is. the best syllable units are selected from the audio-visual corpus and each visual information of selected syllable units are concatenated. There are two processes to obtain the best units. One is to select the N-best candidates for each syllable. The other is to select the best smooth unit sequences, which is done by Viterbi decoding algorithm. For these process, the two distance proposed between syllable units. They are a phonetic environment distance measure and a prosody distance measure. Computer simulation results showed that our proposed algorithm had good performances. Especially, it was shown that pitch and intensity information is also important as like duration information in lip sync.