• Transactions of the Korean Society of Mechanical Engineers
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• v.17 no.5
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• pp.1054-1065
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• 1993

A computer simulation is performed on the effectiveness of the active minimization of harmonically excited enclosed sound fields for producing global reduction in the amplitude of the pressure fluctuations. In this study for the appreciable reductions in total time averaged acoustic potential energy, $E_{pp}$, the transducer location strategies for three dimensional active noise control is presented based on a state space modal which approximates the closed acoustic field.In this study, the above theoretical basis is used to investigate the application of active control to sound fields of low modal density. By the used of room-like 3-dimensional rectangular enclosure it is demonstrated that the reductions in $E_{pp}$ can be achieved by using a single secondary source, provided that the source is placed within the half a wavelength from the primary source and placed away from nodal line of the sound field. Concerning the reductions in $E_{pp}$ by minimzing the pressure in sound fields by the use of 3-dimensional rectangular enclosure, the effects of the number of sensors and the locations of these sensors are investigated. When a few modes dominate the response it is found that if only a limited number of sensors are located away from nodal line and located at the pressure maxima of the sound field such as at each corner of a rectangular enclosure.

• Journal of Broadcast Engineering
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• v.19 no.6
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• pp.819-825
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• 2014

In this paper, monaural ambient component extraction algorithm based on nonnegative matrix factorization (NMF) is described. The ambience component extraction algorithm in this paper is developed for audio upmixing system; Recent researches have shown that they can enhance listener envelopment if the extracted ambient signal is applied into the multichannel audio upmixing system. However, the conventional method stores all of the audio signal and processes all at once, so it cannot be applied to streaming system and digital signal processor (DSP) system. In this paper, the ambient component extraction algorithm based on on-line nonnegative matrix factorization is developed and evaluated to solve the problem. As a result of analysis of the processed signal with spectral flatness measures in the experiment, it was shown that the developed system can extract the ambient signal similarly with the conventional batch process system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2C
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• pp.119-125
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• 2011

Start This paper analyzes the propagation characteristics of acoustic signals in indoor environments, which is applicable to indoor positioning system. Indoor stereo sound system is generally valid within $25m^2$. So it is not possible to apply prevalent sound propagation characteristic to indoor positioning system because the prevalent acoustic signals propagation characteristic is defined under free space condition. Therefore, in this paper, we present the propagation characteristics of acoustic signals in indoor environments considering the free space propagation characteristic as well as room characteristic such as humidity, temperature, absorption of atmosphere and so on. To verify the designed propagation model of indoor acoustic signals, this paper presents the propagation characteristics of decreasing sinusoidal signals whose frequencies ate from 1kHz to 20kHz in anechoic room. In addition, this paper also presents the propagation characteristics of decreasing sinusoidal signals which have multiple frequencies.

• Composites Research
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• v.18 no.5
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• pp.15-20
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• 2005

In this paper, damage induced acoustic emission in the composite plate in numerically simulated by using the three dimensional finite element method and explicit time integration. Acoustic source is modeled by equivalent volume source. To verify the proposed method, dynamic displacements due to the elastic wave are compared with the experiment when the fiber is broken in the single fiber embedded isotropic plate. For the laminated composite plates, the results are compared between homogenized model and DNS approach which models fibers and matrix separately. To capture high frequencies in the elastic wave, small time step size and a large number of meshes are required. The parallel computing technology is introduced to solve a large scale problem efficiently.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.269-272
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• 2004

주행 상태에서의 자동차 내부 음 환경은 다양한 소음 및 구조적 요인으로 인하여 음성에 대한 정상적인 전송 특성을 갖기 어렵다. 이는 음원으로부터 음성 입력 장치(Microphone)에 이르기까지의 채널 왜곡에 기인한 문제로써, 실제 주행 환경에서의 음성 인식 성능에 대해서도 심각한 악영향을 초래한다. 본 논문에서는 주행 소음의 크기에 따른 채널별 음성 왜곡 정도에 따른 명료도를 음성 전달 지수인 STI(Speech Transmission Index)를 통하여 분석하고 그 결과를 음성 인식률과 상호 비교하였다. 그리고 수음 패턴에 따른 명료도 척도와 음성 인식 성능과의 상관성을 검토하고, 이를 통해 단일 채널 환경에서 최적의 마이크로폰 위치에 대하여 고찰해 보았다. 실험 결과, 주행 중의 소음 환경에서도 음성의 명료도 척도와 인식률과의 관계는 높은 상관성이 얻어짐을 알 수 있었고, 각 채널 간의 성능 편차 패턴도 주행 환경에 따라 비슷한 양상을 보이는 것으로 나타났다.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.1815-1816
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• 2006

소음은 환경 오염원의 하나로서 사람에게 육체적, 정신적 피해를 발생시킨다. 이에 소음 제어 기술의 필요성이 증대하였고, 그 중에서 제어가 까다롭고 고비용을 요구하는 저주파 소음 제어 기술의 개발이 확대되고 있다. 따라서 본 연구에서는 저주파의 주기적인 특성을 가지는 1차원 평면파 소음에 대한 능동 제어를 위해 기존의 적응 피드포워드 방법의 단점을 보완하는 적응 피드백 방법을 이용한 능동 소음 제어 시스템을 구성하고 능동 소음 제어 실험을 수행하였다. 이를 위해 소음원과 제어 음원을 가지는 덕트 형상의 실험 장치를 구성하였다. 제어기 설계를 위해 전파 소음을 예측하는 선형 예측법을 적용한 적응 디지털 필터를 구성하였으며 적응 알고리즘으로 Filtered-X LMS 알고리즘을 이용하였다. 제어기는 제어 알고리즘을 프로그램화하여 DSP에 입력함으로써 구성하였다. 실험에 사용된 소음은 500[Hz] 이하의 단일 주파수의 정현파 소음을 사용하였으며, 실험결과 음압 감소의 효과를 볼 수 있었다. 능동 소음 제어의 기술을 개발하여 하드웨어(덕트)의 모양 및 구조, 제어기의 종류 및 처리 속도, 주파수나 크기와 같은 특성이 급격히 변하는 소음의 경우에 능동적으로 소음을 제어할 수 있으며, 저주파 소음을 발생시키는 관형 연소기와 같은 장치 및 여러 분야에 응용이 가능하도록 하였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.82-86
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• 2001

In this Paper, we Propose synthetic aperture technique for twin-line may. Sin91e-line way is required long aperture size in order to achieve high SNR and angular resolution in shallow water Ultra low frequency signal from far-field has left-right ambiguity at sing1e-line array. To resolve these Problems, we'd like to adopt the synthetic aperture technique to twin-line array. The synthetic aperture method adopts coherent processing of sub-aperture signals at successive tine intervals in the beam domain. The proposed method shows low nile error and improved angular resolution. In simulation result, average sidelobe level is reduced about 7〔dB〕when the array Peformed 5-synthesis.

• The Journal of the Acoustical Society of Korea
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• v.39 no.2
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• pp.120-130
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• 2020

This paper solves the problem in underdetermined convolutive mixture by improving the disadvantages of the multichannel nonnegative matrix factorization technique widely used in blind source separation. In conventional researches based on Spatial Covariance Matrix (SCM), each element composed of values such as power gain of single channel and correlation tends to degrade the quality of the separated sources due to high variance. In this paper, level and frequency normalization is performed to effectively cluster the estimated sources. Therefore, we propose a novel SCM and an effective distance function for cluster pairs. In this paper, the proposed SCM is used for the initialization of the spatial model and used for hierarchical agglomerative clustering in the bottom-up approach. The proposed algorithm was experimented using the 'Signal Separation Evaluation Campaign 2008 development dataset'. As a result, the improvement in most of the performance indicators was confirmed by utilizing the 'Blind Source Separation Eval toolbox', an objective source separation quality verification tool, and especially the performance superiority of the typical SDR of 1 dB to 3.5 dB was verified.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.11
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• pp.156-163
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• 2015

In the using of FBG(Fiber Bragg Grating) developed in home land, we designed and manufactured united FBG acoustic transducers the first in Korea. they are being applied to multi-point signal detection of FBG Hydrophone used Hopper WDM(national patent NO 10-1502954) in the underwater. On united FBG transducers manufactured, we made an demonstrated on respective frequency response peculiarities in the underwater and analyzed the special characters. As the experimental result on frequency response peculiarities, we made it possible underwater acoustic detection on united FBG acoustic transducers type to maximum 30Hz~2.5KHz. it's the optimum conditions of 1.2KHz frequency in detection. And for the purpose of realization on multi-point signal detection on wide scope in the underwater, in the using of WDM(Wavelength Division Multiplexing) method and passive band-pass filter system, established arrays system and succeeded in multi-point underwater acoustic signal detection to the frequency 200Hz~1.3KHz out of the two united type FBG transducers. Additionally, it would be possible directivity detection for the object of its source as the intensity of detection signal varies with the sound source's direction and angle. From now on we prepared a new moment on the practical use study on FBG hydrophone in the future.

• Journal of Advanced Navigation Technology
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• v.14 no.4
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• pp.571-577
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• 2010

Driving aircraft requires extremely complicated and detailed information processing. Pilots perform their tasks by selecting the information relevant to them. In this processing, spatial information presented simultaneously through crossmodal link is advantageous over the one provided in singular sensory mode. In this paper, probability to apply providing visual spatial information along with auditory information to enemy tracking system in aircraft navigation is empirically investigated. The result shows that auditory spatial information, which is virtually created through HRTF is advantageous to visual spatial information alone in attention processing. The findings suggest auditory spatial information along with visual one can be presented through crossmodal link by utilizing stereophonic sound such as HRTF. which is available in the existing simple stereo system.