• The Journal of the Acoustical Society of Korea
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• 제26권3E호
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• pp.87-96
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• 2007

In this paper we propose a new noise reduction algorithm for stationary and nonstationary noisy environments. Our algorithm classifies the speech and noise signal contributions in time-frequency bins, and is not based on a spectral algorithm or a minimum statistics approach. It relies on calculating the ratio of the standard deviation of the noisy power spectrum in time-frequency bins to its normalized time-frequency average. We show that good quality can be achieved for enhancement speech signal by choosing appropriate values for ${\delta}_t\;and\;{\delta}_f$. The proposed method greatly reduces the noise while providing enhanced speech with lower residual noise and somewhat higher mean opinion score (MOS), background intrusiveness (BAK) and signal distortion (SIG) scores than conventional methods.

• International Journal of Control, Automation, and Systems
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• 제6권6호
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• pp.818-827
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• 2008

In this paper, we propose a new noise estimation and reduction algorithm for stationary and nonstationary noisy environments. This approach uses an algorithm that classifies the speech and noise signal contributions in time-frequency bins. It relies on the ratio of the normalized standard deviation of the noisy power spectrum in time-frequency bins to its average. If the ratio is greater than an adaptive estimator, speech is considered to be present. The propose method uses an auto control parameter for an adaptive estimator to work well in highly nonstationary noisy environments. The auto control parameter is controlled by a linear function using a posteriori signal to noise ratio(SNR) according to the increase or the decrease of the noise level. The estimated clean speech power spectrum is obtained by a modified gain function and the updated noisy power spectrum of the time-frequency bin. This new algorithm has the advantages of much more simplicity and light computational load for estimating the stationary and nonstationary noise environments. The proposed algorithm is superior to conventional methods. To evaluate the algorithm's performance, we test it using the NOIZEUS database, and use the segment signal-to-noise ratio(SNR) and ITU-T P.835 as evaluation criteria.

• Current Optics and Photonics
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• 제2권5호
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• pp.413-421
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• 2018

We propose a novel methodology based on the multiplication function to improve the signal-to-noise ratio (SNR) for vibration detection in a phi optical time-domain reflectometer system (phi-OTDR). The extreme-mean complementary empirical mode decomposition (ECEMD) is designed to break down the original signal into a set of inherent mode functions (IMFs). The multiplication function in terms of selected IMFs is used to determine a vibration's position. By this method, the SNR of a phi-OTDR system is enhanced by several orders of magnitude. Simulations and experiments applying the method to real data prove the validity of the proposed approach.

• 대한원격탐사학회지
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• 제33권2호
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• pp.159-169
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• 2017

위성용 전자광학탑재체는 제한된 소모전력 및 우주방사선과 같은 사용 환경에 의해 설계부터 특별한 요구사항을 가지고 있으며, 획득 영상의 품질은 주로 GSD (Ground Sampled Distance), 신호대잡음비(SNR, Signal to Noise Ratio), MTF (Modulation Transfer Function)에 따라 좌우된다. 영상센서의 출력신호에 포함된 잡음 감소를 통한 신호대잡음비 개선을 위하여, 센서에 추가된 프리픽셀(Pre-pixel) 및 다크픽셀(Dark-pixel)을 사용하여 CDS (Corrective Double Sampling) 방식을 통해 영상센서의 잡음 성분을 포함한 오프셋 신호(Offset Signal)를 제거하는 아날로그 신호처리(ASP, Analog Signal Processor) 방법을 제안한다. 또한 센서 제어시스템에서는 영상의 불균일성 처리를 위해 제어시스템의 출력 포트별 게인(Gain), 오프셋, 및 센서의 화소별 특성을 반영한 다양한 방식에 의한 보정 방법이 적용된다. 본 논문에서는 이상 설명한 여러 가지 잡음 개선방법을 시스템 설계 및 운영에 적용하여 위성탑재용 전자광학카메라의 신호대잡음비 향상 방법을 제안하고, 실험을 통해 검증한다.

• Journal of Astronomy and Space Sciences
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• 제35권4호
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• pp.279-286
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• 2018

Radar sensors are used for space situational awareness (SSA) to determine collision risk and detect re-entry of space objects. The capability of SSA radar system includes radar sensitivity such as the detectable radar cross-section as a function of range and tracking capability to indicate tracking time and measurement errors. The time duration of the target staying in a range cell is short; therefore, the signal-to-noise ratio cannot be improved through the pulse integration method used in pulse-Doppler signal processing. In this study, a method of improving the signal-to-noise ratio during range migration is presented. The improved detection performance from signal processing gains realized in this study can be used as a basis for comprehensively designing an SSA radar system.

• 전자공학회논문지
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• 제51권9호
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• pp.165-170
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• 2014

음성 인식을 위한 전처리기로 주변 잡음을 제거해 주는 음성향상 기법이 강조되고 있다. 다양한 음성향상 기법들 중 코드북 기반 음성향상 기법은 nonstationary 잡음 환경에서도 효율적으로 동작한다. 하지만, 기존 코드북 기반 음성향상 기법에서는 입력 신호와 음성 및 잡음 코드벡터 간에 미스매치가 발생하여 부정확한 게인이 추정되는 문제가 있다. 본 논문에서는 부정확한 게인을 보상하기 위해 long-term 잡음 추정 알고리즘을 사용하여 매 프레임 별로 신호 대 잡음비기반의 Normalized Weighting Factor (NWF)를 구하고, 이것을 기존 게인에 보상하는 방식을 제안한다. 제안된 코드북 기반 음성향상 기법은 기존 코드북 기반 음성향상 기법에 비해 향상된 성능을 보였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2000년도 하계학술대회 논문집 D
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• pp.2921-2923
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• 2000

Adaptive template filtering has been proposed recently for an enhancement of signal-to-noise ratio. In some magnetic resonance images whose gray levels have relatively small dynamic ranges, e.g., T1 imaging, however, artificial stair-like artifact is observed in edge regions. This is partially due to edge enhancement effect in such voxels that contain multiple compounds at the boundaries of tissues. The gray levels of these voxels tend to change those of near voxels that contain single compound by the adaptive filtering, which exaggerate edge discontinuities. In this paper, we propose a technique to eliminate such artifact by identifying those voxels and assigning a larger template for them. Filtered images with the proposed technique show substantial visual enhancement at the edges without degradation of peak signal-to-noise ratio compared to the original adaptive template filtering for both magnetic resonance images and phantom images

• 한국전자통신학회논문지
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• 제14권6호
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• pp.1249-1256
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• 2019

본 논문에서는 보청기의 음향궤환 및 잡음을 제거하기 위한 새로운 알고리즘을 제안한다. 이 알고리즘은 기존의 FIR 구조를 이용하는 대신 신경망 적응예측필터를 이용한 심층학습 알고리즘으로 궤환 및 잡음제거 성능을 향상시킨다. 먼저 궤환제거기가 마이크 신호에서 궤환신호를 제거하고, 이어서 Wiener 필터기법을 이용하여 잡음을 제거한다. 잡음 제거는 음성신호가 가진 주기적 성질에 따라 선형예측모델을 이용하여 잡음이 포함된 음성신호로부터 음성을 추정해내는 것이다. 한 루프 안에 포함된 두 적응 시스템의 안정적 수렴을 보장하기 위해 궤환제거기 및 잡음제거기의 계수 업데이트를 분리하여 실시하며 제거 후 생성된 잔차신호를 이용하여 수렴시키는 과정을 진행한다. 본 연구에서 제안한 궤환 및 잡음제거기의 성능을 검증하기 위하여 시뮬레이션 프로그램을 작성하고 모의실험을 수행하였다. 실험 결과, 제안한 심층학습 알고리즘을 사용하면 기존의 FIR 구조를 사용하는 경우보다 궤환제거기에서 약 10 dB의 SFR(: Signal to Feedback Ratio), 잡음제거기에서 약 3 dB의 SNRE(: Signal to Noise Ratio Enhancement) 개선효과를 얻을 수 있는 것으로 확인되었다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1985년도 학술발표회 논문집
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• pp.87-94
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• 1985

In this paper, enhancement of speech corrupted by additive white or colored noise is stuided. The nuconstrained frequency-domain block least-mean-square (UFBLMS) adaptation algorithm and its frequency-weighted version are newly applied to speech enhancement. For enhancement of speech degraded by white noise, the performance of the UFBLMS algorithm is superior to the spectral subtraction method or Wiener filtering technique by more than 3 dB in segmented frequency-weighted signal-to-noise ratio(FWSNERSEG) when SNR of speech is in the range of 0 to 10 dB. As for enhancement of noisy speech corrupted by colored noise, the UFBLMS algorithm is superior to that of the spectral subtraction method by about 3 to 5 dB in FWSNRSEG. Also, it yields better performance by about 2 dB in FWSNR and FWSNRSEG than that of time-domain least-mean-square (TLMS) adaptive prediction filter(APF). In view of the computational complexity and performance improvement in speech quality and intelligibility, the frequency-weighted UFBLMS algorithm appears to yield the best performance among various algorithms in enhancing noisy speech corrupted by white or colored noise.

• 대한전기학회논문지:시스템및제어부문D
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• 제53권2호
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• pp.118-126
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• 2004

The modified Wiener filtering method is proposed for effective noise suppression in edge region of images corrupted by additive white gaussian noise. Although the pixels classified as a edge region in the conventional Wiener filter have lots of noise components, the conventional Wiener filter cannot remove noise effectively due to the preserving of edges. To reduce noise well in edge region, we modify filter coefficients of the conventional Wiener filter. The modified filter coefficients increase in noise suppression effect in edge region, while they preserve edges for strong edge region. From simulation (256${\times}$256 size, 256 graylevel images) filtered images by the proposed method show much improved subjective image quality with higher peak signal-to-noise ratio compared to those by the conventional Wiener filtering.