• 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.

• 대한전자공학회논문지
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• 제22권2호
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• pp.60-65
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• 1985

A method of weighting of a staggered pulse train is presented and an improvement per-formance of the signal-to-noise ratio for each case is compared. As the result of a computer simulation, the signal-to-noise ratio for weighting of a stagger-ed pulse train is a great improvement on the case of an uniform pulse train. The signal-to-noise ratio of optimum weighting is more improved than that of binomial weighting, it is known that the signal-to-noise ratio is improved as the increasing of the number of delay line cancellers, and it is shown that the signal-to-noise ratio is improved by weighting of the MTI output pulses before the integration process.

• 대한전자공학회논문지
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• 제22권1호
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• pp.41-45
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• 1985

디지탈 이동물체지시기 계통(DMTI System)에서 신호대잡음비(SNR)의 저하는 적정처리 전에 디지탈 이동물체지시기의 출력펄스를 weightlng시킴에 의하여 개선될 수 있구 #정상살기의 수가 증가함에 따라 신호대잡음비가 개선됨을 나타내었다. Optimum weighting으로 얻어진 신호대잡음비는 binomial weighting으로 얻어진 것보다 더 커짐을 알았다. 컴퓨터 시뮬레이션 결과를 통하여 각각의 경우에 대한 신호대잡음비의 배선특성을 나타낸다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 춘계학술대회논문집
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• pp.495-498
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• 2004

In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined by a simple sound insulation measurement.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.441-444
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• 2004

In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined through field measurements.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.260-263
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• 2000

A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

• 대한기계학회논문집A
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• 제21권8호
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• pp.1332-1338
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• 1997

In the case of the transfer function for the secondary path is dependent on time, the on-line method which can model it is continuously must be applied to the active noise control technique. And the adaptive random noise technique among the on-line methods is effective in the narrow-band control. In this method, the signal to noise ratio between random noise for modeling and primary noise is low. Therefore, the estimations of transfer function will be prone to inaccuracies and the convergence time will be too long. Such imperfections will have an influence upon the performance of an active noise controller. In this study, t enhance the signal to noise ratio, the on-line method that is combined the conventional adaptive random noise technique and the adaptive line enhancer, is proposed. By using proposed on-line method, a rigorous system identification and control of primary noise have been implemented.

• Archives of Pharmacal Research
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• 제10권1호
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• pp.1-8
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• 1987

One of the problems of derivatie spectrophotometry, the decrease of signal-to-noise ratio by derivative operations, was solved by three concepts of digital filtering, ensemble averaging, least squares polynomial smoothing and Fourier smoothing. The suthors made several compouter programs written in APPLE SOFT BASIC language for the actual applications of the concepts of these digital filters on UV spectrophotometer system. As a result, ensemble averaging could not be used as a routine operation for the spectrophotometer used. The maximum S/N ratio enhancement factors achieved by least squares polynomial smoothing were 6.17 and 7.47 for the spectra of Gaussian and Lorentzian distribution models, and by Fourier smoothing 16.42 and 11.78 for the spectra of two models, respectively.

• 전자공학회논문지A
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• 제30A권12호
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• pp.98-107
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• 1993

This paper describes a design of a multi-bit oversampling noise-shaping D/A converter which achieves a resolution of 12 bits using oversampling technique. In the architecture the essential block which determines the whole accuracy is the analog internal D/A converter, and the designed charge-integration internal D/A converter adopts a differential structure in order to minimize the reduction of the resolution due to process variation. As the proposed circuit is driven by signal clocks which contains the information of the data variation from the noise-shaping coder, it minimizes the disadvantage of a charge-integration circuit in the time axis. In order to verify the circuit, it was integrated with the active area of 950$\times$650${\mu}m^{2}$ in a double metal 1.5-$\mu$m CMOS process, and testified that it can achieve a S/N ratio of 75 dB and a S/(N+D) ratio of 60 dB for the signal bandwidth of 9.6 kHz by the measurement with a spectrum analyzer.

• 한국음향학회지
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• 제16권2호
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• pp.5-9
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• 1997

본 논문에서는 chirp 신호를 초음파 신호처리 시스템에 입력시킬 때 반사신호의 진폭을 최대로 얻기위한 chirp 신호의 대역폭을 결정하는 방법을 연구하였다. 초음파 신호처리 시스템에서 반사신호의 S/N 비는 초음파의 전파시에 발생하는 감쇠와 산란에 의하여 매우 낮게된다. Chirp 신호를 사용한 펄스 압축기술은 초음파 반사시스템의 S/N 비를 증가시킬 수 있는 방법중 하나이다. 시뮬레이션 및 실험결과, chirp 신호의 펄스폭이 넓을수록 출력이 증가하였고 동일한 펄스폭과 주어진 초음파시스템의 대역폭에 대한 최대출력은 chirp 신호의 대역폭이 초음파시스템의 대역폭 보다 1.15배 클 때 얻어짐을 확인하였다.