• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.45-55
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• 1999

In this paper, we used the WLMS(Wavelet domain Least Mean Square) adaptive filter based on the wavelet transform to cancel grain noise. Usually, grain noise occurs in changes of the crystalline structure of metals in high temperature environment. It makes the detection of flaw difficult. The WLMS adaptive filtering algorithm establishes the faster convergence rate by orthogonalizaing the input vector of adaptive filter as compared with that of LMS adaptive filtering algorithm in time domain. We implemented the WLMS adaptive filter by using the delayed version of the primary input vector as the reference input vector and then implemented the CA-CFAR(Cell Averaging- Constant False Alarm Rate) threshold estimator. CA-CFAR threshold estimator enables to detect the flaw and back echo signals automatically. Here, we used the output signals of adaptive filter as its input signal. To Cow the statistical characteristic of ultrasonic signals corrupted by grain noise, we performed run test. The results showed that ultrasonic signals are nonstationary signal, that is, signals whose statistical properties vary with time. The performance of each filter is appreciated by the signal-to-noise ratio. After LMS adaptive filtering in time domain, SNR improves to about 2-3㏈ but after WLMS adaptive filtering in wavelet domain, SNR improves to about 4-6㏈.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.3
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• pp.203-208
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• 2010

In this paper, we convert underwater acoustic signal made by sonar system into digital image. We propose the algorithm that detects target candidate and emphasizes information of target introducing image processing technique for the digital image. The process detecting underwater target estimates background noise in underwater acoustic signal changing irregularly, recomposes it. and eliminates background from original image. Therefore, it generates initial target group. Also, it generates weighted map through proceeding doppler information, ensures information for target candidate through filtering using weighted map for image eliminated background noise, and decides the target candidate area in the single frame. In this paper, we verified that proposed algorithm almost had eliminated the noise generated irregularly in underwater acoustic signal made by simulation, targets had been displayed more surely in the image of underwater acoustic signal through filtering and process of target detection.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.181-184
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• 1998

웨이브렛 변환은 신호나 영상을 분석하기 위한 다해상도 분해기법으로 사용되어 왔다. 웨이브렛 변환영역에서 신호는 스케일과 위치상의 크기로 표현된다. 이 변환영역에서는 신호나 영상의 주파수 성분들이 각각의 스케일에 따라서 분리되어 나타난다. 또한 각 변환영역은 신호나 영상의 공간적인 특성을 상당부분 포함하고 있다. 이러한 웨이브렛 변환의 특성은 푸리에 변화에 기초한 방법과는 달리, 에지와 잡음성분을 효과적으로 분리할 수 있는 정보를 우리에게 제공해 준다. 본 논문에서는 웨이브렛 변환영역의 각 스케일 특성과 공간적인 특성을 이용하여 영상의 잡음성분을 제거하였다. 잡음제거 기법의 성능평가를 위해 Wiener 필터링 방법과 비교하였다.

• The Journal of the Acoustical Society of Korea
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• v.34 no.6
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• pp.493-498
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• 2015

Complex acoustic signals can be formed in a water-filled acoustic pulse tube under some exciting conditions. It makes difficult to measure plane-wave reflection coefficient with the pulse tube for low frequency bands. In this study, using COMSOL Multiphysics we show that the tube wall excitation generates complex acoustic field of nonplanar mode as well as planar one. From such field incident or reflected planar mode can be decomposed respectively with a modal decomposition method, two-dimensional Fourier filtering. It makes possible to more accurately determine the plane-wave reflection coefficient of acoustic specimen with time gating.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.54-62
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• 2000

Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.42-47
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• 1999

In this paper, a mixture interacting multiple model (MIMM) algorithm is proposed to enhance speech contaminated by additive nonstationary noise. In this approach, a mixture hidden filter model (HFM) is used to model the clean speech and the noise process is modeled by a single hidden filter. The MIMM algorithm, however. needs large computation time because it is a recursive method based on multiple Kalman filters with mixture HFM. Thereby, a computationally efficient implementation of the algorithm is developed by exploiting the structure of the Kalman filtering equation. The simulation results show that the proposed method offers performance gain compared to the previous results in [4,5] with slightly increased complexity.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.185-188
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• 2000

본 논문은 인간이 양귀로 3차원 공간상의 음의 위치를 파악하는 원리를 이용하여 만들어진 입체음향을 청취자의 안귀에 가장 효과적으로 재생시킴으로써 완벽한 입체음향을 재현하는 장치로 3차원 게임, 가상현실 시뮬레이터, 놀이공원에서 사용될 수 있는 A/V용 입체음향 재생기 구현 기법이다. 본 개발 품은 전방 30도에 스피커를 배치하는 기존 제품에 비해 안정된 정면 음상 정위와 후면 음상 정위가 가능하고, 청취자의 머리 움직임에 따른 입체음향 효과의 저하를 막을 수 있으며, 인지적 특성을 고려한 역 필터링으로 과도한 신호처리로 인한 음질 저하를 개선할 수 있고, 다중 사용자에게 동일한 음질을 골고루 전달할 수 있어 카 오디오나 영화관에서도 사용되어질 수 있다.

• Transactions of the Korean Society of Mechanical Engineers
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• v.15 no.3
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• pp.809-823
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• 1991

본 연구에서는 Wiener 필터링 이론에 의하여 소음원의 입력신호에 대한 최적 한 예측을 할 수 있는 최적예측기(optimal predictor)로써 부가적인 음을 발생시키고 입력신호 및 출력신호 간의 차인 오차를 최소화시키도록 하는 적응신호처리방법에 대 해 설명하고 이러한 적응 신호처리 방법을 이용한 능동 소음 제어 방법을 제시하였다. 이와 아울러 제어계의 환경 변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해 야 하는 적응 소음 제어계에서, 음향궤환과 함께 필히 고려해야하는 부가적인 전달함 수-모델과 스피커를 포함하는 보조경로 및 오차미이크로폰을 포함하는 오차경로의 전 달함수의 영향을 고려한 능동소음제어에 대해 연구하였다.

• The Magazine of the IEIE
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• v.31 no.6
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• pp.17-39
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• 2004

본 논문에서는 다채널 오디오 시스템에서 사용되어지는 음향신호처리에 대하여 저자들이 그간 제안하였던 방법들을 중심으로 다뤘다. 다룬 내용은 첫째로 각 스피커에서 청자의 위치까지의 공간응답의 영향을 배재할 수 있는 역 필터링이고, 둘째로 다채널 스피커의 위치를 파악하여 최적 위치와의 차이를 자동적으로 보상할 수 있는 방법이다. 셋째로 인간의 인지적인 측면을 고려하는 다채널 스피커로부터의 에너지 레벨 정렬 방식에 대하여 다뤘고, 마지막으로 특정공간의 반사음의 분포 패턴을 구하여 이를 일반적인 청취공간에서 재현해 내는 음장 재현 방식에 대하여 요약하였다.(중략)

• The Journal of the Acoustical Society of Korea
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• v.31 no.8
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• pp.532-541
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• 2012

An underwater transient signal is distinguished from an ambient noise. Database for the underwater transient signal is required since the underwater transient signal shows various characteristics depending on acoustic features. In the paper, hence, sound mask-filter was applied to extract the transient signals which exist temporally and locally in the ocean. The standard signal was chosen and cross-correlated with the raw signal. A mask-filter for a transient signal was obtained using the threshold which was decided by the maximum likelihood method in the envelope of the cross-correlated signal. Using the sound mask-filter, the transient signal of a sea catfish {Galeichthys felis (Linnaeus)} was extracted from the underwater ambient noise. Similarly, the man-made signal was added into the noise and it was extracted by the same method. We also have demonstrated the significance of the transient signal through comparing the extracted signals depending on the standard signal. In the results, the proposed method, sound mask-filtering, could be utilized as a database construction of the transient signals in underwater noise. Particularly, this study would be useful to extract the wanted signal from arbitrary signals.