• Nuclear Engineering and Technology
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• 제39권6호
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• pp.717-724
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• 2007

A measurement-based time-domain noise simulation of radiation detector-preamplifier (front-end) noise in nuclear spectroscopy is described. The time-domain noise simulation was performed by generating "noise random numbers" using Monte Carlo's inverse method. The probability of unpredictable noise was derived from the empirical cumulative distribution function via the sampled noise, which was measured from a preamplifier output. Results of the simulated noise were investigated as functions of time, frequency, and statistical domains. Noise behavior was evaluated using the signal wave-shaping function, and was compared with the actual noise. Similarities between the response characteristics of the simulated and the actual preamplifier output noises were found. The simulated noise and the computed nuclear pulse signal were also combined to generate a simulated preamplifier output signal. Such simulated output signals could be used in nuclear spectroscopy to determine energy resolution degradation from front-end noise effect.

• 한국안광학회지
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• 제17권2호
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• pp.203-207
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• 2012

목적: 다른 생체신호와 마찬가지로 망막전도(electroretinogram, ERG) 신호도 측정시 잡음이 발생한다. 이 잡음을 효과적으로 제거하여 망막관련 진단의 정확도를 높이고자 하였다. 방법: ERG 신호에 60 Hz 잡음과 백색잡음을 발생시켜 샘플링 신호를 만들었다. 웨이브렛 변환과 대역통과 필터를 이용하여 잡음를 제거하였다. 푸리에 변환 스펙트럼을 이용하여 제거된 주파수를 비교하였다. 신호대잡음비(signal to noise ratio, SNR)를 이용하여 제거된 잡음을 수치적으로 비교하였다. 결과: 푸리에 변환 스펙트럼을 비교한 결과 웨이브렛 변환에서는 60 Hz 잡음은 완전히 제거 되었으며 백색잡음도 많이 제거되었다. 대역통과필터에서는 60 Hz와 백색잡음 남아 있었다. 신호대잡음비를 비교한 결과에서는 웨이브렛 변환은 22.8638, 대역통과 필터는 4.0961로 나타났다. 결론: 웨이브렛 변환을 이용하여 잡음 제거시 신호의 왜곡을 적게 발생시켜 제거할 수 있었다. 망막전도 신호를 이용한 망막 진단에 정확도를 높일 수 있을 것으로 기대된다.

• 한국음향학회지
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• 제18권6호
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• pp.3-8
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• 1999

본 연구에서는 신호원에 대한 사전 정보 없이 혼합된 신호로부터 잡음 신호만을 선택적으로 제거할 수 있는 새로운 형태의 능동 소음 제거기(Active noise canceller)를 제안한다. 음성신호와 같은 동특성을 갖는 신호의 분리에 효과적으로 사용되는 동적 재귀 신경망(Dynamic recurrent neural network)을 원하는 신호원에 섞인 잡음신호를 분리하여 선택적으로 제거하기 위한 능동 소음 제거기의 전처리기로 이용한다. 능동 소음 제거기는 분리된 잡음 신호에 대한 역위상 신호를 적응적으로 발생함으로써 특정 위치에서 원하는 신호만을 선택적으로 남길 수 있도록 한다. 컴퓨터 시뮬레이션을 통하여 제안된 시스템이 선택적인 소음제거에 효과적임을 보인다.

• Ocean and Polar Research
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• 제28권4호
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• pp.459-465
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• 2006

Ambient noise as a background noise in the ocean has been well known for its the various and irregular signal characteristics. Generally, these signals we treated as noise and they are analyzed through stochastical level if they don't include definite sinusoidal signals. This study is to see how ocean ambient noise can be analyzed by the chaotic analysis technique. The chaotic analysis is carried out with underwater ambient noise obtained in areas near the Korean Peninsula. The calculated physical parameters of time series signal are as follows: histogram, self-correlation coefficient, delay time, frequency spectrum, sonogram, return map, embedding dimension, correlation dimension, Lyapunov exponent, etc. We investigate the chaotic pattern of noises from these parameters. From the embedding dimensions of underwater noises, the assesment of underwater noise by chaotic analysis shows similar results if they don't include a definite sinusoidal signal. However, the values of Lyapunov exponent (divergence exponent) are smaller than that of random noise signal. As a result we confirm the possibility of classification of underwater noise using Lyapunov analysis.

• 센서학회지
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• 제20권6호
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• pp.412-419
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• 2011

In this paper, we used the inter-bit noise detection method in order to reduce the effects of noise light in a visible light identification system that uses a visible LED as a carrier source. A visible light identification system consists of a reader and a transponder. When the enable signal from the reader is detected, the transponder encodes the response data in RZ(Return-to-Zero) bit stream and sends response signal by modulating a visible LED. The reader detects the response signal mixed with noise light, samples the noise voltage in each blank low time between data bits of the RZ signal, and recovers the original data by subtracting the sampled noise from the received signal. In experiments, we improved the signal-to-noise ratio by 20dB using the inter-bit noise detection method.

• 대한전자공학회논문지
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• 제13권5호
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• pp.24-29
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• 1976

Phase locked AM신호 검파방식에서는 위상고정회로를 사용하여 입력신호로부터 반송신호를 분리 재생시킨다. 입력잡음은 백색 Gaussian잡음이고, 전려제어발진기의 자유발진주파수와 입력반송신호주파수가 같다는 가정하에 위상고정회로의 동작특성을 해석하고, 본 검파방식의 신호대 잡음비를 정량적으로 고찰하였다. Phase locked AM신호 검파방식은 종래의 검파방식에 비해서 잡음의 영향을 적게 받게됨을 본 해석에서 알 수 있다. In the phase locked AM signal detection, phase locked loop is used to extract a synchronous carrier from an input AM signal. Under the assumption that input noise is white Gaussian and free-running frequency of voltage controlled oscillator is the same that of an input carrier, operational behaviours of phase locked loop is analyzed and signal to noise ratio of the detection is derived quentitatively. The results show that the phase locked AM signal detection method offers a higher degree of noise mmunity than conventional AM signal detections.

• 전기학회논문지P
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• 제62권4호
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• pp.192-197
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• 2013

Road traffic noise barriers being used to reduce the noise, but the city surroundings inhibition, ecosystem disturbance, and it is difficult to maintain. Can enhance or complement the existing noise barrier performance, so that it is necessary to develop an electronic noise-reduction system In this paper, we proposed an electronic road noise reduction devices to reduce road noise for a DSP-based signal processing and analog signal input-output controller. In order to verify the control performance, we performed noise reduction experimentation of ANC by filtered-X LMS algorithm and traffic noise signal injection. The controller is equipped with noise reduction algorithms were tested on the characteristics of directional and omnidirectional speaker.

• 한국멀티미디어학회논문지
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• 제19권2호
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• pp.128-137
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• 2016

In mobile communication, air conduction(AC) speech signal had been commonly used, but it was easily affected by ambient noise environment such as emergency, military action and rescue. To overcome the weakness of the AC speech signal, bone conduction(BC) speech signal have been used. The BC speech signal is transmitted through bone vibration, so it is affected less by the background noise. In this paper, we proposed noise cancellation algorithm of the BC speech signal using noise feature of decomposed bands. The proposed algorithm consist of three steps. First, the BC speech signal is divided into 17 bands using perceptual wavelet packet decomposition. Second, threshold is calculated by noise feature during short time of separated-band and compared to absolute average of the signal frame. Therefore, the speech and noise parts are detected. Last, the detected noise parts are removed and then, noise eliminated bands are re-synthesised. In order to confirm the efficiency of the proposed algorithm, we compared the proposed algorithm with conventional algorithm. And the proposed algorithm has better performance than the conventional algorithm.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 추계종합학술대회 논문집
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• pp.1183-1186
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• 1998

Adaptive nois canceller can extract the noiseremoved spech in noisy speech signal by adapting the filter-coefficients to the background noise environment. A kind of LMS algorithm is one of the most popular adaptive algorithm for noise cancellation due to low complexity, good numerical property and the merit of easy implementation. However there is the matter of increasing misadjustment at voiced speech signal. Therefore the demanded speech signal may be extracted. In this paper, we propose a fast and noise robust wavelet packet adaptive noise canceller with NLMS-SUM method LMS combined algorithm. That is, we decompose the frequency of noisy speech signal at the base of the proposed analysis tree structure. NLMS algorithm in low frequency band can efficiently dliminate the effect of the low frequency noise and SUM method LMS algorithm at each high frequency band can remove the high frequency nosie. The proposed wavelet packet adaptive noise canceller is enhanced the more in SNR and according to Itakura-Satio(IS) distance, it is closer to the clean speech signal than any other previous adaptive noise canceller.

• 전기학회논문지
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• 제60권1호
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• pp.201-204
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• 2011

For efficient noise reduction in a Fourier transform ion cyclotron resonance (FT-ICR) mass spectrum, a new algorithm was proposed. The suggested algorithm reduces white and electrical noise, and it improves signal-to-noise ratio. This algorithm has been optimized to reduce the noise more efficiently using the traces of signal level. The algorithm has been efficiently combined with derivative window to improve the resolution as well S/N. Time domain data was corrected for DC voltage interference. $t^n$ window was applied in time domain data to improved the resolution. However, $t^n$ window can improve the signal resolution, it will also increase the noise level in frequency domain. Therefore, newly developed noise reduction algorithm will be applied to make a balance between resolving power and S/N ratio for magnitude mode. The trace algorithm can determine the current data point with several data points (mean, past data, calculated past data). In the current calculations, we assumed data points with S/N ratio more than 3 were considered as signal data points. After the windowing and noise reduction, both resolution and signal-to-noise ratio were improved. This algorithm is applicable more efficiently to frequency dependent noise and large size data.