• 대한전자공학회논문지SP
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• 제41권3호
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• pp.231-238
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• 2004

주파수 영역에서의 적응 신호처리는 입력의 자기 상관 행렬에 이산 퓨리에 변환(DFT Discrete Fourier Transform)을 이용할 때 거의 대각선화 되는 특성으로 인해 시간영역 적응필터보다 주파수 영역 적응 필터가 빠르게 적응한다. 본 논문에서는 변형된 이산 퓨리에 변환(MDFT: modified DFT)을 이용하여 주파수 영역 적응 필터를 설계함으로써 안정한 수렴 속도를 갖는 잡음 제거 시스템을 제안한다. 제안한 구조는 MDFT를 이용하여 연산수를 최소화하며, 안정한 수렴을 유지하면서 블록 없는 처리를 할 수 있고, 최적의 수렴 속도를 위해 입력 자기 상관 행렬에 MDFT를 사용해 근사적으로 대각화 시키고 시간적으로 변하는 스텝 크기를 정규화 하는 고속 적응 잡음 제거(HANR: high speed adaptive noise reduction) 알고리즘이다. HANR 알고리즘을 적용한 필터는 DFT변환법을 사용한 LMS방법(non-proposed)보다 30%정도의 속도 개선이 있다.

• 한국소음진동공학회논문집
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• 제18권3호
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• pp.284-292
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• 2008

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, FXLMS algorithm has the demerit that stability of the control is decreased when the step size become larger but the convergence speed is faster because the step size of FXLMS algorithm is fixed. As a result, the system has higher probability which the divergence occurs. Thus the Co-FXLMS algorithm was developed 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 and experimental results show that active noise control using Co-FXLMS is effective in reducing the noise in duct system.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 춘계학술대회논문집
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• pp.598-603
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• 2007

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, FXLMS algorithm has the demerit that stability of the control is decreased when the step size become larger but the convergence speed is faster because the step size of FXLMS algorithm is fixed. As a result, the system has higher probability which the divergence occurs. Thus the Co-FXLMS algorithm was developed 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 the noise in duct system.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.234-237
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• 2000

A number of theoretical researches have been done in recent years on the restoration of images and a variety of algorithms have been developed to implement noise reduction methods. However the blurring effect has not been perfectly overcome in the process of noise reduction. In this paper, we propose a new approach to image restoration that the blurring effect is significantly decreased and the performance of the noise reduction improves by eliminating the noise in the wavelet transform domain in comparison with the conventional noise reduction methods. The proposed algorithm performs much better than the conventional in the subjective image quality and PSNR performance. It is verified through computer simulations,

• 한국정보전자통신기술학회논문지
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• 제6권1호
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• pp.23-28
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• 2013

본 논문에서는 모바일 환경에서 dual microphone을 이용하여 배경 잡음을 효율적으로 제거하는 방법을 제안한다. 참조 마이크와 주 입력 마이크로부터 측정된 신호를 각각 FDAF의 참조신호와 주 입력 신호로 하여 잡음 제거를 수행한 뒤에 MMSE-LSA를 이용하여 잔여 잡음 및 배경 잡음을 추정하여 제거 한다. 이때 일관성 있는 잡음 제거 성능을 위해 두 마이크 신호의 PLD를 이용한 VAD 결과를 사용한다.

• 오토저널
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• 제14권5호
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.

• 전기학회논문지
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• 제61권10호
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• pp.1489-1495
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• 2012

In this study, the active noise control experiment has been performed using induction motor noises. While the noises were measured, a induction motor was operated in different speed. For the simulation of ANC(Active Noise Control), test-bed is composed a multi-channel ANC system was constructed. In order to compare the control performance, we performed noise reduction simulations of ANC by Co-FXLMS algorithm and FXLMS algorithm. Through the simulation results, we confirmed that convergence performance and noise decrease effect of the proposed Co-FXLMS algorithm have been improved from existing FXLMS algorithm.

• 대한의용생체공학회:의공학회지
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• 제28권6호
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• pp.732-743
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• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.

• Journal of Electrical Engineering and Technology
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• 제10권5호
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• pp.2197-2204
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• 2015

In recent years, medical image diagnosis has growing significant momentous in the medicinal field. Brain and lung image of patient are distorted with salt and pepper noise is caused by moving the head and chest during scanning process of patients. Reconstruction of these images is a most significant field of diagnostic evaluation and is produced clearly through techniques such as linear or non-linear filtering. However, restored images are produced with smaller amount of noise reduction in the presence of huge magnitude of salt and pepper noises. To eliminate the high density of salt and pepper noises from the reproduction of images, a new efficient fuzzy based median filtering algorithm with a moderate elapsed time is proposed in this paper. Reproduction image results show enhanced performance for the proposed algorithm over other available noise reduction filtering techniques in terms of peak signal -to -noise ratio (PSNR), mean square error (MSE), root mean square error (RMSE), mean absolute error (MAE), image enhancement factor (IMF) and structural similarity (SSIM) value when tested on different medical images like magnetic resonance imaging (MRI) and computer tomography (CT) scan brain image and CT scan lung image. The introduced algorithm is switching filter that recognize the noise pixels and then corrects them by using median filter with fuzzy two-sided π- membership function for extracting the local information.

• 응용통계연구
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• 제24권5호
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• pp.905-914
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• 2011

영상처리 분야에서 원래의 순수 이미지를 오염시키는 잡음을 제거하는 것은 매우 중요한 문제이다. 그러나 여러 가지 원인으로 인하여 발생하는 잡음의 발생을 완벽하게 막는 것은 현실적으로 불가능하다. 따라서 영상에 포함되어 있는 잡음을 제거하거나 최대한 줄이는 것이 매우 중요한 과제이다. 본 연구에서는 이미지를 오염시키고 있는 잡음의 상대적인 크기를 측정하여 잡음의 분산의 수준을 추정하고, 이를 영상처리 분야에서 자주 사용되는 잡음제거 기법인 시그마 필터에 응용하여 잡음을 효과적으로 제거하는 통계적 알고리즘을 제시하였다. 결론적으로, 잡음의 분산의 수준에 관계없이 본 연구에서 제안한 통계적 잡음제거 방법론을 통해 기존의 시그마 필터보다 현저하게 개선된 결과를 얻을 수 있었다.