• 전기의세계
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• v.46 no.5
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• pp.39-44
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• 1997

현재 DSP는 음성 및 오디오 신호처리 시스템, 디지털 통신 시스템, 제어 시스템, 영상처리 시스템 등 많은 영역에 걸쳐 성공적으로 사용되고 있다. 몇가지 대표적인 활용분야를 살펴보면, 음성신호 압축 분야 [1-4], MPEG (moving picture expert group)과 같은 오디오신호 압축분야[5,6], 그리고 디지털 통신 시스템에서의 적응 반향제거기, 적응 동화기, 채널간섭 제거, 변복조기, 채널 코딩, 암호화기[7-14] 등에서도 DSP가 사용되고 있다. 그리고 수중 음향 신호처리[15], 디지털 필터 디자인, 전력 스펙트럼 추정, 수중 음향 신호처리 같은 디지털 신호처리 분야[16-23]와 적응 신호처리[24-26], 이외에도 능동 소음 제어기 및 적응 제어기와 같은 제어 시스템 [27]에도 유용하게 이용되고 있다. 또한 영상 압축, 디지털 방송, 의료기기 등과 같은 영상처리 분야[28-32] 및 그 밖의 많은 분야에서 DSP의 활용은 점점 커져가고 있는 추세이다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.17-21
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• 2001

There are some methods to enhance the ‘sweet spot’in loudspeaker-based 3D audio systems. Most of them can be only applied to narrow frequency band inherently. In this paper, we introduce the more robust 3D sound reproduction system which has far wider robust bandwidth. The system applies a sum and difference filter to the conventional three loudspeaker-based one.

• 전자공학회논문지 IE
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• v.44 no.4
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• pp.1-10
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• 2007

In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.351-359
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• 2003

In this paper, we proposed a method for acoustic echo cancellation based on independent component analysis. When the large acoustic noise is picked up by the microphone, the performance of echo cancellation decreased. We used two microphones that received echo signal which is linearly mixed with the noise, then separated the echo signals from the received signals with independent component analysis algorithm. The separated echo signal is used for the reference signal of adaptive algorithm which leads to better performance of the echo cancellation. Computer simulation results show the validity of the proposed method.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.213-216
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• 2002

본 논문은 원거리 회의 시스템이나 차량 내 핸즈프리 통화 시 필연적으로 발생하는 음향 반향을 제거하기 위해 새로운 알고리즘을 제안한다. 본 논문에서 제안된 음향반향 제거 시스템은 Delayless 서브밴드 음향반향 제거기와 협대역 동시통화 검출기로 구성된다. Delayless Subband 적응 음향 반향 제거기는 적은 계산 량과 높은 수렴속도로 음향 반향 제거 성능이 뛰어난 것으로 알려져 있으며 본 논문은 이를 이용해 안정적인 음향 반향 제거를 위해서 협대역 Subband 내에서 동시통화 검출기를 구현한다. 기존의 광대역 동시 통화 검출기에 비해 본 논문에서 제안된 협대역 동시통화 검출기는 저주파 Subband 대역에서만 동시통화 검출을 수행하여 down-sampling으로 인한 계산 량 감소와 저주파 특성을 가지는 Subband 대역의 신호 특성으로 인한 신뢰성 있는 통화 상태 정보를 제공함으로서 전체적인 음향반향제어 시스템의 성능을 향상시킬 수 있도록 하였다. 본 연구에서 제안된 음향반향 제어 시스템의 성능은 다양한 컴퓨터 시뮬레이션을 통하여 입증하도록 한다.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2002.05c
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• pp.237-241
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• 2002

전력선과 조명등을 이용한 음향전달 시스템을 구성하여 어학실습 교육용으로의 활용 가능성을 평가하였다. 음성신호를 변조하여 전력선에 실어 보내는 송신기와 조명등의 빛으로부터 음성신호를 검출하여 이어폰 또는 헤드폰 둥으로 들을 수 있는 수신기로 구성하였다. 제작한 송신기와 수신기의 주파수 특성과 스펙트럼분석을 하였다. 스피커로 출력되는 음질상태는 최적상태는 아니지만 음성이나 노래 등을 이용한 어학실습 교육용으로는 충분한 수준이 었다.

• Journal of the Korean Society for Nondestructive Testing
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• v.23 no.3
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• pp.254-262
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• 2003

In this study, acoustic emission (AE) signals due to surface cracking and moisture movement in the flat-sawn boards of oak (Quercus Variablilis) during drying under the ambient conditions were analyzed and classified using the principal component analysis. The AE signals corresponding to surface cracking showed higher in peak amplitude and peak frequency, and shorter in rise time than those corresponding to moisture movement. To reduce the multicollinearity among AE features and to extract the significant AE parameters, correlation analysis was performed. Over 99% of the variance of AE parameters could be accounted for by the first to the fourth principal components. The classification feasibility and success rate were investigated in terms of two statistical classifiers having six independent variables (AE parameters) and six principal components. As a result, the statistical classifier having AE parameters showed the success rate of 70.0%. The statistical classifier having principal components showed the success rate of 87.5% which was considerably than that of the statistical classifier having AE parameters.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.425-433
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• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• The Journal of the Acoustical Society of Korea
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• v.12 no.3
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• pp.49-55
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• 1993

이 논문에서는 알려진 신호를 검파하는 국소 최적 퍼지 검파기의 검정 통계량의 확률 분포를 정확한 방법과 근사적 방법으로 얻었다. 또한, 컴퓨터 모의 실험으로 국소 최적 퍼지 검파기의 성능 특성과 국소 최적 검파기의 성능 특성을 견주어 보았다. 마지막으로 국소 최적 퍼지 검파기의 기본적인 기능적 특성에 대해서도 알아보았다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.2
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• pp.237-245
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• 2012

This paper describes an acoustic echo canceller with double-talk by the blind signal separation. The acoustic echo canceller is deteriorated or diverged in the double-talk period. So we use the blind signal separation to estimate the near-end speech signal and to eliminate the estimated signal from the residual signal. The blind signal separation extracts the near-end signal with dual microphones by the iterative computations using the 2nd order statistical character. Because the mixture model of blind signal separation is multi-channel in the closed reverberation environment, we used the copied coefficients of echo canceller without computing the separation coefficients. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and then operates stably in the normal state without the divergence of coefficients after ending the double-talking. And it shows the ERLE of averagely 20dB higher than the normal LMS algorithm.