• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 제13회 신호처리 합동 학술대회 논문집
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• pp.519-522
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• 2000

Digital watermarking has been proposed as a solution to the problem of copyright protection of the multimedia documents. In this paper a new watermarking method for digital images operating in the frequency domain is proposed. In our approach, DCT coefficients of the watermark are added to the low frequency region of the host image, and extract it using the Wiener Filter. Due to the characteristic of the wiener filtering, the watermark is robust to various image processing techniques. Experimental results show that it is possible to reliably extract the watermark without degrading image quality.

• The Journal of the Acoustical Society of Korea
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• 제25권2E호
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• pp.36-42
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• 2006

In this paper, a novel post processing using Wiener filtering technique is proposed to p rm further interference reduction in FDICA. Using the proposed method, the target signal components are remained with little attenuation while the interference components are drastically suppressed. The results of experiments show that the proposed method achieves a reduction of the residual crosstalk. Compared to the NLMS method, the proposed method has slightly better separation performance in SIR, and even requires much less computational complexity.

• 한국소음진동공학회논문집
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• 제22권12호
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• pp.1227-1236
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• 2012

The low speed machinery faults are usually caused by the bearing failure of the rolling elements. As the life time of the bearing is limited, the condition monitoring of bearing is very important to maintain the continuous operation without failures. A few monitoring techniques using time domain, frequency domain and fuzzy neural network vibration analysis are introduced to detect and diagnose the faults of the low speed machinery. This paper presents a method of fault detection for the rolling element bearing in the low speed machinery using the Wiener filtering and shock pulse counting techniques. Wiener filter is used for noise cancellation and it clearly makes the shock pulse emerge from the time signal with the high level of noise. The shock pulse counting is used to determine the various faults obviously from the shock signal with transient pulses not related with the bearing fault. Machine fault simulator is used for the experimental measurement in order to verify this technique is the powerful tool for the low speed machine compared with the frequency analysis. The test results show that the method proposed is very effective parameter even for the signal with high contaminated noise, speed variation and very low energy. The presented method shows the optimal tool for the condition monitoring purpose to detect the various bearing fault with high accuracy.

• 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.

• 방송공학회논문지
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• 제18권3호
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• pp.455-462
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• 2013

본 논문에서는 여러 개의 마이크를 이용하여 잡음을 제거하는 방법인 공간 필터로 전처리된 신호를 입력으로 하는 음성 왜곡 가중 다채널 위너 필터 (Spatially Preprocessed Speech Distortion Weighted Multi-channel Wiener Filter: SP-SDW-MWF)에 대해 소개하고, 가중치를 결정하는 방법을 제안한다. SP-SDW-MWF는 마이크로폰 어레이를 이용한 잡음 제거 알고리즘으로서 마이크로폰 불일치와 같은 오차에 강인한 것으로 알려져 있다. SP-SDW-MWF는 필터 계수를 최적화할 때 음성 왜곡과 잡음 제거에 대한 기준으로 나누어 가중치를 두고 있다. 이러한 가중치를 결정하기 위해, 본 논문에서는 전력 스펙트럼 밀도 오차를 평가 척도로 사용하여 마이크로폰으로부터 입력된 음성 신호와 잡음의 전력 스펙트럼 밀도의 비 (a priori SNR)를 이용하는 방법을 제안한다. 실험결과에서 나타난 바와 같이 a priori SNR에 따라 가변적인 가중치를 사용하는 것이 고정된 값을 가중치로 사용하는 것보다 향상된 성능을 보임을 알 수 있다.

• 천문학회지
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• 제27권2호
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• pp.191-201
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• 1994

We propose to use the entropy of power spectra defined in the frequency domain for the deconvolution of extended images. Spatial correlations requisite for extended sources may be insured by increasing the role of power entropy because the power is just a representation of spatial correlations in the frequency domain. We have derived a semi-analytical solution which is found to severely reduce computing time compared with other iteration schemes. Even though the solution is very similar to the well-known Wiener filter, the regularizingng term in the new expression is so insensitive to the noise characteristics as to assure a stable solution. Applications have been made to the IRAS $60{\mu}m\;and\;100{\mu}m$ images of the dark cloud B34 and the optical CCD image of a solar active region containing a circular sunspot and a small pore.

• 대한전자공학회논문지SP
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• 제46권5호
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• pp.162-166
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• 2009

본 논문은 적응 반향 제거기인 parametric wiener filter (PWF)의 반향신호 제거 파라미터를 구간 조건에 따라 차등적으로 적용시키는 새로운 음향학적 반향제거 (Acoustic Echo Suppression, AES) 알고리즘을 제안한다. PWF는 반향 신호 추정의 부정확성을 보상하고자 반향신호 제거 파라미터를 사용한다. 기존의 방법은 단일통화 구간과 동시통화 구간에 관계없이 동일한 반향신호 제거파라미터 값을 적용하여 동시통화 환경에서 근단 신호의 왜곡을 발생 시킨다. 본 논문에서는 상관계수 (Correlation Coefficient)를 이용한 동시통화(Double-Talk) 검출 알고리즘과 음성구간검출기 (Voice Activity Detector, VAD)의 결과로부터, 원단 화자만 있는 구간과 동시통화 구간을 구분하고 각각의 경우에 반향제거 파라미터를 달리 적용하여 동일한 반향신호 제거 파라미터를 사용함에 따른 동시통화 구간내의 근단 화자신호의 음질 저하 없이 반향신호를 보다 효과적으로 제거하는 방법을 제안한다. 제안된 방법의 결과는 객관적인 실험을 통해 비교 평가한 결과 우수한 성능을 보였다.

• 한국전자통신학회논문지
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• 제11권1호
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• pp.45-52
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• 2016

본 논문은 영상 복원 문제에 대한 정규화 모수를 찾는 새로운 방법을 제시한다. 위너 필터(Wiener filter)는 원본 영상과 잡음의 파워 스펙트럼 등의 사전 정보를 요구한다. 제약된 최소자승 복원 역시 노이즈 수준에 대한 지식을 요구한다. 사전 정보가 없으면 티코노프(Tikhonov) 정규화 모수를 선택하기 위한 일반화된 교차 검증법이나 L자형 곡선 검정 등의 별도의 최적화 함수가 필요하다. 본 논문에서는 주파수 영역에서 선형 시스템의 바이어스 항목과 티코노프 정규화 시스템의 평활화 항목을 연결하는 자기 정규화 방법을 제안하고 영상 복원 문제에 적용한다. 실험결과는 제안하는 방법의 효능을 보여준다.

• The Journal of the Acoustical Society of Korea
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• 제23권2E호
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• pp.56-67
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• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• 대한음성학회지:말소리
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• 제48호
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• pp.155-173
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• 2003

This paper proposes a noise reduction-based speech detection method under telephone channel environments. We adopt the AURORA front-end noise reduction algorithm based on the two-stage mel-warped Wiener filter approach as a preprocessor for the frequency domain speech detector. The speech detector utilizes mel filter-bank based useful band energies as its feature parameters. The preprocessor firstly removes the adverse noise components on the incoming noisy speech signals and the speech detector at the next stage detects proper speech regions for the noise-reduced speech signals. Experimental results show that the proposed noise reduction-based speech detection method is very effective in improving not only the performance of the speech detector but also that of the subsequent speech recognizer.