• 한국전자파학회논문지
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• 제25권4호
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• pp.418-425
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• 2014

본 논문에서는 원거리에서 사람의 호흡 및 심박수를 측정할 수 있는 Ku-Band 대역의 도플러 레이더 시스템을 설계하였다. 레이더로 측정을 하는 동시에 ECG(Electrocardiogram)를 이용하여 호흡 및 심박수 측정을 하였다. 수신된 I/Q(Inphase & Quadrature-phase) 신호를 이용한 arctangent demodulation을 적용하였을 때 RMSE(Root Mean Square Error)값을 약 50 % 정도 개선시킬 수 있었다. CW(Continuous Wave) 도플러 레이더는 송수신단이 연속적으로 항상 열려 있기 때문에 송신안테나에서 수신 안테나로 누설되는 전력이 발생하게 된다. 출력 파워가 높아질수록 누설되는 전력은 시스템의 SNR(Signal-to-Noise Ratio)에 영향을 주게 된다. 따라서 본 연구에서는 원거리에서 감지가 가능하도록 누설 전력의 반대위상을 갖는 신호를 더해줌으로써 누설 전력으로 인한 수신단의 감도 저하를 최소화 시키는 기법을 적용하였다. 위의 leakage cancellation 기법을 적용하여 최대 35 m 거리 대상의 호흡 및 심박 수를 측정하였으며, 이와 비교하여 ECG 값에서 추출한 심박 수 값과 일치함을 볼 수 있다.

• The Journal of the Acoustical Society of Korea
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• 제21권4E호
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• pp.178-182
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• 2002

In this paper, we propose an adaptive wavelet method for car interior noise cancellation. For this purpose, we use a node dependent threshold which minimizes the Bayesian risk. We propose a noise estimation method based on spectral entropy using histogram of intensity and a candidate best basis instead of Donoho's best bases. And we modify the hard threshold function. Experimental results show that the proposed algorithm is more efficient, especially to heavy noisy signal than conventional one.

• 조명전기설비학회논문지
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• 제13권3호
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• pp.17-24
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• 1999

본 연구는 스피터와 마이크폰쌍을 이용하여 변압기 소음 감소를 위한 적응 능동소음제어에 있어서 스피커-증폭기-마이크로폰 경로와 스피거-마이크로폰 쌍의 동특성에 대한 이론적인 내용과 시뮬레이션을 통하여 이를 확인하였다. 또한 음향경로 내에 존재하는 마이크로폰-스피커 쌍의 전달함수를 SLS(sequential least square)알고리즘으로 추정하였으며, 추정된 전달함수에 대한 identify는 z 평면에서 안정된 극점과 영점을 갖고 있음을 확인하였다.

• 한국통신학회논문지
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• 제21권8호
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• pp.2043-2054
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• 1996

In this paper, Digital Delay Locked Loop(DDLL) is designed, implemented and analysed by experiment whose results show that it is possible to track the received signal by this scheme. Designed digital DLL has an advantage that it is not needed to maintain gain balance between early and late channels, which has been problem with an analog DLL. Also DDLL has more improved noise performance compared to analog DLL due to noise level limitation and noise cancellation characteristics. For various loop parameters, their effects on loop performance are analysed and simulated. Proposed DDLL is the first attempt as a digital approach in code tracking loop and it is expected to be a good reference for spread spectrum communication research.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2005년도 하계학술대회
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• pp.397-405
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• 2005

This paper presents a headphone system which has adopted two algorithm to increase sound clearness and to separate signal from noisy environment. In the field of adaptive signal processing, LMS algorithm which is a kind of steepest decent method, can be implemented with more simple calculation, so that we use it to eliminate unwanted noise elements for the proposed system. Futhermore we generate early echo using some delays, then mix it in signal. This process can increase the clearness of signal. In this paper, we prove that the proposed system can be implemented in real time. The proposed system is satisfied to subject assessment test base on MOS(Mean Opinion Score) of ITU-T.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1996년도 춘계학술대회
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• pp.143-146
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• 1996

An adaptive digital filtering of the electroencophalogram(EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, an artifact which not be acceptable in some cases of automatic EEG processing. The types of electrical artifact seen on EEG records is described. Those are the EOG and the PVC roller pump noise. And we study the method for cancelling these artifacts. The method does not need the reference channel, and are obtained by cascading the linear predictor and the noise canceller. The simulation results illustrate the performances of the presented method in terms of the capability of interferences suppression.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.802-805
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• 1994

There are strong ocean wave interference with big amplitude very low frenqencies are similar to the ship's hydrodynamic signals. To detect ship's hydrodynamic field will be faced various natural hydrodynamic interferences which are radom and the prior knowlege of which are not know. This paper proposes to use the adaptive noise cancelling principle and used the model of adaptive wave canceller to eliminate the ocean wave interfrence and detect the ship's hydrodynamic signals. Computer simulation results shown that signal to noise ratio can be raised from several to ten times. It shows the fact that this mathod can detect the ship's hydrodynamic signals from the strong ocean wave interferences while it is difficult for the old methods.

• 한국소음진동공학회논문집
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• 제20권8호
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• pp.734-741
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• 2010

In this paper, an intake RCV system for low noise turbo engine was developed through optimization process of a geometric path of compressor housing and an open rate of recirculation valve. At first, the critical customer requirement from voice of customer was defined and quality function deployment of an intake RCV system was executed. And then, the renovative concept design using pugh matrix method was selected as final concept for satisfaction of requirement. Simultaneously, system analysis was carried by function diagram and fishbone diagram. Next, control factors and levels for the optimal design were performed. And, the optimal design of an intake RCV system was studied using design of experiment. Conclusively, we achieved not only cancellation tip-out noise at the driving condition but also improvement of NVH commodity through optimization process of an intake RCV system, which is optimal configuration of compressor housing and recirculation valve.

• 전력전자학회논문지
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• 제2권2호
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• pp.35-40
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• 1997

히스테리시스 제어방식은 전력변환장치나 모터구동시스템의 전류제어를 위하여 흔히 사용된다. 이 제어방식은 에러 신호가 미리 설정된 밴드폭을 초과하려고 하는 순간마다 제어신호를 바꿔줌으로써 측정된 신호가 기준신호를 추종하도록 하는데, 본 논문에서는 이와 같은 빠른 응답특성을 갖는 히스테리 제어방식을 가청소음을 억제하기 위한 능동 소음제거에 적용하였다. 펜티엄 프로세서와 사운드블러스터 16이 소음 제어 실험을 위하여 사용되었으며, 사운드블러스터 16은 아날로그/디지탈 또는 디지털/아날로그 변환을 수행할 뿐만 아니라 가청 소음 저감을 위한 상쇄원 스피커의 구동원으로 사용된다.

• 전기학회논문지
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• 제64권12호
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• pp.1756-1760
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• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.