• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.42-46
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• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.32-39
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• 1994

The speech transmission quality in telephone is quantified in terms of loudness rating, but this method has been validated only for the handset telephony. The transmission quality of loudspeaking telephony in any room must be evaluated not only with speech transmission but also with background noise, echo and reverberation since the effect of room acoustics is much stroger for loudspeaking telephoy. Therefore, it requires a better approach to specify the quality of loudspeaking telephony. By calcuating the speech transmission index (STI), a physical method for measuring the quality of speech transmission was proposed by Steeneken. In this paper, the application of a STI method for estimating articulation of loudspeaking telephony was discussed. And the STI measurement system with high speed calculation was also three rooms, having different reverberation times. The result show that the STI decreases as the reverberation time of rooms increases. It suggests that speech transmission index method can be useful evaluating articulation of a loudspeaking telephony including the sound field characteristics.

• The Journal of the Acoustical Society of Korea
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• v.25 no.8
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• pp.395-402
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• 2006

When an active SONAR works in the subsurface. its detection Performance is limited by the reverberation. The winds Play a primary role in the Production of bubbles in the ocean. And the bubbles as efficient scatters contribute to the reverberant field. In this Paper the effects of wind-generated bubbles on sound propagation in the subsurface are investigated as a mid-frequency Hull-mounted SONAR works. The active signal excess is calculated at source depths 3. 5. and 10m considering bubble layer for frequencies 5. 7.5, and 10kHz. The change of the near-surface sound speed tend to increase surface reverberation levels and change the active signal excess. In the 10m/s winds. the maximum detection range reduces over 3km through the near-surface . The reason is the upper refraction due to the wind-generated bubbles.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.10a
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• pp.42-45
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• 1996

잔향시간이 긴 실내의 전달함수의 모델링에 있어서 , 일반적인 플 밴드 MA 모델에 기초한 적응 필터를 이용한 경우에는, 많은 필터 차수를 필요로 하고 적응 시간이 길어지는 문제점이 있다. 본 논문에서는 필터 차수를 감소시키고 수렴 특성을 향사시키기 위해서, 각 입출력 신호를 몇 개의 주파수 대역으로 나우어서 각각의 주파수 대역에 대새서 적응 처리 과정을 행하는 서브밴드 MA 모델을 이용한 적응디지털 필터 처리 방법을 제안한다. 컴퓨터 시뮬레이션 서브밴드MA 모델을 이용한 디지털 적응 필터 처리과정의 유효성을 나타냈었다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.71-77
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• 2005

In this paper we implement a headphone-based 5.1 channel 3-dimensional (3D) sound generation system using HRTF (Head Related Transfer Function). Each mono sound source in the 5.1 channel signal is localized on its virtual location by binaural filtering with corresponding HRTFs, and reverberation effect is added for spatialization. To reduce the computational burden, we reduce the number of taps in the HRTF impulse response and model the early reverberation effect with several tens of impulses extracted from the whole impulse sequences. We modified the spectrum of HRTF by weighing the difference of front-back spec01m to reduce the front-back confusion caused by non-individualized HRTF DB. In informal listening test we can confirm that the implemented 3D sound system generates live and rich 3D sound compared with simple stereo or 2 channel down mixing.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.115-128
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• 2012

This paper investigates the effects of diffuser on the acoustical parameters in music hall with consideration of the result of scattering coefficient measurement. A scale model hall of 600 seats with orchestra shell was used for experiments. The materials of 1:50 scale model was chosen through absorption coefficient measurement based on ISO 354. The model was matched to the computer simulation model in terms of reverberation time. In order to evaluate the effect of diffuser location, the measurements were accomplished with and without diffusers according to 7 configurations by diffuser-installed region; sidewall, balcony front, ceiling and so on. The following acoustical parameters were extracted from each measurement case; Reverberation time (RT), Early decay time (EDT), Clarity (C80), Center time (Ts), Sound strength (G) and Temporal diffusion (TD) from the auto-correlation function (ACF) of impulse responses. As a result, the absorption power and diffusion power were increased with number of diffusers. Accordingly RT, EDT and G were decreased by diffuser and the redirection of reflections was occurred briskly. Averaged TD was 6.05 to 6.30 by measurement cases. RT was found to be the most related factor to diffusion power (R = 0.94). The correlation between TD and EDT was high (R = 0.73). In addition, the effects of diffuser-installed location were discussed in terms of acoustical parameter variation.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.11b
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• pp.886-889
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• 2002

In order to evaluate absorption coefficient, experimental works for normal and random incident absorption coefficient are made. An investigation for correlation between normal and random incident absorption was carried out by experiment and analysis. It appears that at the low frequencies, the random incident absorption is higher than the normal one, whileas at the high frequencies, the random incident absorption is decreased due to the effect of grazing incident components.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.93-100
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• 1999

확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

• The Journal of the Acoustical Society of Korea
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• v.40 no.6
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• pp.615-625
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• 2021

Prior to proposing appropriate standard for subway station platform, this study conducted field measurements to examine characteristics of room acoustics on platform of two subway stations. As a result of analyzing the longitudinal length of the platform, Sound Pressure Level (SPL) decreased (maximum difference : 14 dB), Reverberation Time (RT) tended to increase (maximum difference of 0.8 s ~ 1.5 s), and C50 and D50 were decreased (maximum difference: 5.9 dB ~ 9.1 dB and 31.8 % ~ 37.6 %, respectively) as measurement positions moved away from the sound source. The Interaural Cross-correlation Coefficient (IACC) did not show clear tendency, but it was lower than 0.3 in entire points. It is judged that the subway platform has non-uniform sound field characteristics due to various combinations of direct and reflective sound even though it is finished with a strong reflective material.This indicates that the room acoustic characteristics of the near and far sound field are clearly expressed depending on the source-receiver distances in the subway platform having a long flat shape with a low height compared to the length.Therefore, detailed architectural and electric acoustic design based on the characteristics of each location of speaker and sound receiver in the platform is required for an acoustic design with clear sound information at all positions of the platform.