• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1845-1851
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• 2007

Background noise and room reverberation are two causes of degradation in speech in listening situations. Many algorithms developed to enhance reverberant speech. In this paper we propose a dereverberation method for enhancement of speech using modified the linear prediction(LP) residual in reverberant room condition. The proposed dereberberation method based on the fact that the signification excitation of the vocal tract system takes place at the instant of glottal closure in voiced speech. Our method used delay information form each sensor, and we need reverberant signals from 3 sensors. We obtain a new LP residual signal using modified IP residual combination which derived form weighting of the LP residual and the Hilbert transform of LP residual. The nature of the coherently added Hilbert envelop has several large amplitude spikes because of the effects of noise and reverberation. This residual of the clean speech is used to excite the time-varying all-pole filter to obtain the enhanced speech. We achieved simulation of proposed algorithm for performance analysis in reverberation environment. The proposed algorithm improves substantially the quality of reverberant speech.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.290-301
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• 2019

The proposed algorithm deals with a detection problem of target echo for narrow-band continuous-wave active sonar in the underwater environment in this paper. In the active sonar systems, ping signal emitted for target detection produces a signal that consists of multiple reflections by many scatterers around, which is called reverberation. The proposed algorithm aims to detect the low-Doppler target echo in the reverberant environment. The proposed algorithm estimates the bearing, frequency, and temporal bases based on beamspace-domain multichannel nonnegative matrix factorization. In particular, the bases are divided into two basis groups - the reverberation group and the echo group, then the basis groups are estimated independently. In order to evaluate the proposed algorithm, a simulation with synthesized reverberation was performed. The results show that the proposed algorithm has enhanced performance than the conventional algorithms.

• Journal of KSNVE
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• v.10 no.4
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• pp.559-564
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• 2000

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.487-490
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• 2002

학교 교육의 특성상 많은 부분이 교실에서의 음성정보 전달에 의해 이루어지고 있는 점을 감안하면 바람직한 청취환경의 개선이 검토되어야 한다. 또한 중${\cdot}$ 고등학교의 수학능력시험의 국어, 영어 듣기평가 및 다양한 어학 시험이 시청각 시설을 통해 이루어지고 있는 실정이므로 교실의 음환경은 매우 중요한 요소라하겠다. 본 논문에서는 음환경을 좌우하는 음원의 위치에 따라 명료 도가 어떻게 달라지는지를 실험을 통하여 검증하고, 명료도가 높고, 교실 전체에 균등한 분포를 보이는 음원의 위치를 찾아내고자 하였다. 교실 내의 음원의 위치로는 일반적으로 많이 쓰이고 있는 column(벽면 노출형)과 ceiling(천정 매입형) 위치와 임의의 음원 cluster(전면 중앙)를 선정하여 음장 파라메터를 측정한 결과 RASTI 는 세 타입 모두 $0.54\~0.55$로 값으로 근소한 차이를 보이고 있으며, 잔향시간은 ceiling>cluster>column의 순서로 나타났다. 일반적으로 잔향과 명료도와의 관계는 반비례하는 것으로 알려져 있으나, 실험 결과 잔향시간이 1.33초로 가장 긴 column 스피커의 경우 D50 값이 약 $47\%$로 가장 높은 값으로 나타났다. 이것은 column형 스피커의 경우 음원과 각 학생의 위치에 대한 평균 직접음선거리가 가장 짧기 때문인 것으로 나타났다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.443-446
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• 2000

본 연구는 과학적이고 실험적인 접근을 통하여 공간의 비례와 실내음향과의 상관관계를 규명하기 위하여 체적과 마감재료가 같고 비슷한 기본축척의 모델을 중심으로 가로, 세로, 높이의 비율이 다른 4개의 모델에서 각각 SPL, EDT, C, RT, LEF, IACC 등을 측정하여 각 형태의 모델의 값을 비교 분석하였다. 실험결과 잔향은 동일한 체적인 경우 음원으로부터 평균거리가 멀리 위치한 장방형의 형태가 양쪽 측벽 사이의 거리가 넓은 장방형보다 잔향시간이 긴 것으로 나타났다. 음압은 실의 체적이 작고 음원으로부터의 평균거리가 짧은 순서로 큰 반면, 명료도는 잔향의 크기순서에 반대로 분석되었으며 특히 음원으로부터의 직선거리가 짧은 순서로 명료도가 좋게 나타났다. IACC는 양쪽 측벽의 거리가 가장 긴 모델에서 가장 작게 나타났다 그러나, 초기 음장에서는 그 변화가 작게 나타났는데 이것은 IACC가 실의 형태에 장시간 관여하는 지수라는 것을 의미한다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.411-414
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• 2004

거리종속 해양환경에서 정상모드 모델 중 하나인 KRAKEN 과 Lambert 공식 형태의 산란 함수를 이용하여 양상태 잔향음을 모델링 하였다. 단열 (adiabatic) 정상모드 모델을 이용하여 거리종속 환경에서의 송신기와 산란체, 산란체와 수신기 사이의 음장을 계산하며 각 산란체에서의 산란 강도는 각 모드별 파수에 의한 입사각 및 산란각을 이용, 모드 조합에 의해 계산하였다. 또한 광대역 모델링을 이용하여 시간 영역에서의 잔향음 신호를 계산하였다.

• The Journal of the Acoustical Society of Korea
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• v.7 no.2
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• pp.26-38
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• 1988

This paper describes three methods for the estimation of the impulse reponse in an indoor acoustic transfer system which rather has long reverberation time by the cross spectrum. The first method, which is the conventional one, will use the white noise as the source signal. Therefore, the very long time window data and numerous number of DFT are necessary for this estimation. The second method has been disigned in order to shorten the length of time window of the first method by using a burst of noise as the source signal. The third method which will be suggested in this paper uses too types of definite signal with short duration time of the source signal. According to the view point of computation capacity, and estimation accuracy of the impulse response, the compared experimental results show that the third method will be better than the others.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.94-100
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• 2001

In October 1997, low-frequency reverberation was measured at an experimental site off the west coast of the Jeju island using the explosive charge, Signals Underwater Sound (SUS). Received signals were separated into the noise, the reflection, and the scattering region, and then were analyzed for the spectral and statistical characteristics of each region. In the analysis of the spectrum we verified that each region had a unique frequency band and statistical characteristics as well. The results of this analysis showed that the real and imaginary portions were shown to be both normal distributions in each frequency bin. The reverberation envelope had a Rayleigh distribution and the phase had a uniform distribution.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.3-7
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• 2001

In active sonar system using CW signal, when the noise included reverberation has not the white characteristics, the CFAR detector estimates high threshold. Because of this reason it cannot detect targets and not resolve the closely spaced multiple targets. In order to solve these problems, we propose an adaptive reverberation rejection filter The proposed filter is composed of an adaptive filter and a fixed filter with its coefficients. To study the performance of the proposed adaptive reverberation rejection filter, various experiments have been performed under In moving active sonar environments. As a results, the proposed method has the improved performance than the previous methods.

• Journal of Korean Association for Spatial Structures
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• v.11 no.1
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• pp.67-75
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• 2011

The present work investigate the effect of sound reinforcement systems on the acoustics of a large dome stadium using a computer simulation. The acoustics of a dome stadium was predicted analyzed by varying room absorption, as well as the configuration of speakers including their directivity, installed height, and numbers. It was found that both D50 and RASTI were improved by increasing room absorption. Larger EDTs were observed according to the increase of room absorption at mid and high frequencies. On the other hand, RT did not show any significant correlation with the changes in room absorption, which might be the effect of a forced linear fitting for non -linear energy decay process. With respect to the speaker configuration, the speech intelligibility of a sound reforcement system installed at higher placed more relied upon their directivity rather than room absorption, Also, lower placing of speakers was found to be effective in decreasing RTs regardless of room absorption.