• 한국음향학회지
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• 제33권5호
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• pp.326-334
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• 2014

논문은 MVDR 빔 형성 기법을 위한 새로운 공분산 행렬 예측을 제안한다. 오디오-비디오 센서를 이용하여 음원의 방향 정보를 정확히 추적함으로써, 공분산 행렬은 가변 적응 망각율을 적용하여 효과적으로 예측된다. 가변 적응 망각율은 신호 대 방해 신호 비를 고려하여 결정된다. 실험 결과에서는 제안하는 방법의 성능이 방해신호/잡음 감소 및 음성 왜곡의 면에서 기존의 방법의 성능보다 더 우수하다는 것을 보여준다.

• 설비공학논문집
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• 제3권2호
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• pp.103-113
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• 1991

This paper presents a method for estimating the noise source contribution of a pulse combustion burner in a multiple input system where the input sources may be coherent each other. By coherence function method, it is found that the biggest part of the noise source in the pulse combustion burner is generated by the part of the combustion chamber. This analysis is modeled as three input / single output system because the noise generating mechanism of the pulse combustion burner is very complicated. The coherence function method is proved to be useful tool for the identification of noise source. The overall levels of the radiated source pressure by coherence function method are compared with those measured and calculated by the frequency response function approach. The experimental results have shown a good agreement with the results calculated by the coherence function method when the input sources are coherent strongly each other. The estimation of shield effect by FRF method indicates that significant reduction can be achieved in sound radiation if only transmission path generated by the part of combustion chamber is acoustically shield.

• 음성과학
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• 제11권3호
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• pp.131-142
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• 2004

Speaker identification system basically functions by comparing spectral energy of an individual production model with that of an input signal. This study aimed to develop a new speaker identification system from two parameters from the spectral energy of numeric sounds: difference sum and correlation coefficient. A narrow-band spectrogram yielded more stable spectral energy across time than a wide-band one. In this paper, we collected empirical data from four male speakers and tested the speaker identification system. The subjects produced 18 combinations of three-digit numeric. sounds !en times each. Five productions of each three-digit number were statistically averaged to make a model for each speaker. Then, the remaining five productions were tested on the system. Results showed that when the threshold for the absolute difference sum was set to 1200, all the speakers could not pass the system while everybody could pass if set to 2800. The minimum correlation coefficient to allow all to pass was 0.82 while the coefficient of 0.95 rejected all. Thus, both threshold levels can be adjusted to the need of speaker identification system, which is desirable for further study.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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• pp.737-742
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• 2007

The method of the reducing duct noise can be classified by passive and active control techniques. However, passive control has a limited effect of noise reduction at low frequencies (below 500Hz) and is limited by the space. On the other hand, active control can overcome these passive control limitations. The active control technique mostly uses the Least-Mean-Square (LMS) algorithm, because the LMS algorithm can easily obtain the complex transfer function in real-time particularly when the Filtered-X LMS (FXLMS) algorithm is applied to an active noise control (ANC) system. However, the convergence performance of the LMS algorithm decreases slightly so it may delay the convergence time when the FXLMS algorithm is applied to the active control of duct noise. Thus the Co-FXLMS algorithm was developed to improve the control performance in order to solve this problem. The Co-FXLMS algorithm is realized by using an estimate of the cross correlation between the adaptation error and the filtered input signal to control the step size. In this paper, the performance of the Co-FXLMS algorithm is presented in comparison with the FXLMS algorithm. Simulation results show that active noise control using Co-FXLMS is effective in reducing duct noise.

• 제어로봇시스템학회논문지
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• 제7권8호
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• pp.656-663
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• 2001

This paper presents a sensor fusion method to recognize a cylindrical object a CCD camera, a laser slit beam and ultrasonic sensors on a pan/tilt device. For object recognition with a vision sensor, an active light source projects a stripe pattern of light on the object surface. The 2D image data are transformed into 3D data using the geometry between the camera and the laser slit beam. The ultrasonic sensor uses an ultrasonic transducer array mounted in horizontal direction on the pan/tilt device. The time of flight is estimated by finding the maximum correlation between the received ultrasonic pulse and a set of stored templates - also called a matched filter. The distance of flight is calculated by simply multiplying the time of flight by the speed of sound and the maximum amplitude of the filtered signal is used to determine the face angle to the object. To determine the position and the radius of cylindrical objects, we use a statistical sensor fusion. Experimental results show that the fused data increase the reliability for the object recognition.

• 방송공학회논문지
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• 제18권1호
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• pp.106-114
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• 2013

본 논문에서는 내부 표본화 주파수와 핵심 대역 하향 표본화 비의 조정이 MPEG USAC의 전체 성능에 끼치는 영향을 다룬다. 여기서 내부 표본화 주파수는 실제 부호화될 때의 내부적인 표본화 주파수를 말한다. 핵심 대역 하향 표본화 비는 전체 부호화 대역폭에 대한 핵심 대역폭의 비를 나타낸다. 성능은 6개의 실험 음원에 대해 10명이 참가한 MUSHRA 검사를 통해 평가되었다. 실험에서는 기존의 1/2에 비해 1/3 혹은 1/4의 핵심 대역 하향 표본화 비를 사용하면 저전송률에서 부호화 성능을 향상시킬 수 있음을 보였다.

• 센서학회지
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• 제28권6호
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• pp.371-376
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• 2019

We have developed an acoustic resonance inspection system to inspect the welding defects in the mechanical parts fabricated using friction stir welding method. The inspection system was consisted of a DAQ board, a microphone sensor, an impact hammer, and controlled by a PC software. The system was developed to collect and analyze the sound signal generated by hitting the sample with an impact hammer to determine whether it is defective. In this study, 100% welded good samples were compared with 95%, 90%, and 85% welded samples, respectively. The variation of the completeness in welding did not affect the visual appearance in the samples. As a result of analyzing the natural frequencies of the good samples, the five natural frequency peaks were identified. In the case of the defective samples, the frequency change was observed. The welding failure detection time was fast enough to be only 0.7 seconds. Employing our welding defect inspection system to the actual industrial field will maximize the efficiency of quality inspection and thus improve the productivity.

• 한국위성정보통신학회논문지
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• 제8권4호
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• pp.17-20
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• 2013

본 연구에서는 싸인 보드의 집중도를 올리기 위한 초음파 스피커 시스템의 음성신호이득, 방사패턴과 같은 상태정보 표시를 위한 LED 구동 회로를 소개한다. 초음파 스피커 시스템은 에너지 손실이 적고 음이 멀리 전달된다. 이러한 특징을 지닌 초음파 스피커는 일상생활에서 다양하게 쓰일 수 있다는 점에서 매우 유용하다. 상기 장점을 살려서 제안된 LED 회로는 초음파 스피커 시스템으로부터 인터페이스를 제공받아 선형적인 LED 밝기로써 상태정보를 표현한다. 설계된 회로는 동부 0.35um CMOS 공정을 사용하였다.

• Journal of Astronomy and Space Sciences
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• 제28권1호
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• pp.79-91
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• 2011

Ongnu (Jade Clepsydra; also called Heumgyeonggaknu) is a water clock was made by Jang Yeong-sil in 1438. It is not only an automatic water clock that makes the sound at every hour on the hour by striking bell, drum and gong, but also an astronomical clock that shows the sun's movement over time. Ongnu's power mechanism used is a water-hammering method applied to automatic time-signal device. The appearance of Ongnu is modeled by Gasan (pasted-paper imitation mountain) and Binpungdo (landscape of farming work scene) is drawn at the foot of the mountain. The structure of Ongnu is divided into the top of the mountain, the foot of the mountain and the flatland. There located are sun-movement device, Ongnyeo (jade female immortal; I) and Four gods (shaped of animal-like immortals) at the top of the mountain, Sasin (jack hour) and Musa (warrior) at the foot of the mountain, and Twelve gods, Ongnyeo (II) and Gwanin on the flatland. In this study, we clearly and systematically understood the time-announcing mechanism of each puppet. Also, we showed the working mechanism of the sun-movement device. Finally, we completely established the 3D model of Ongnu based on this study.

• 한국음향학회지
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• 제18권5호
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• pp.63-67
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• 1999

최근 들어 실내음향 특성의 측정과 평가를 위해 백색잡음을 이용한 잔향시간 측정법 대신에 임펄스응답을 이용한 측정법이 널리 이용되고 있다. 이 방법은 재현성이 우수하고 다양한 실내음향 특성치들을 한꺼번에 산출할 수 있어 전통적인 잔향시간 측정법에 비해 여러 가지 장점을 가지고 있다. 본 연구에서는 MLS(Maximum Length Sequence) 신호를 이용하여 실내에서의 임펄스 응답을 측정하고 이를 후처리(post-processing) 하여 잔향시간(EDT, RT), 명료도 지수(C50, C80, D, U50, U80, AI), 음의 크기 지수(G) 등, 주로 실의 음성음향 성능을 측정하는 시스템을 구축하였다. 본 연구에서는 측정시스템과 후처리 프로그램의 구성, 몇몇 실내공간에 대한 시험 측정의 결과 및 고찰 등에 대해 소개하고자 한다.