• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2008년도 춘계종합학술대회 A
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• pp.784-788
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• 2008

In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2001년도 추계학술대회논문집 II
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• pp.1335-1340
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• 2001

The information technology brought us many kinds of conveniences to our life, but it also caused social problems such as privacy interference, unexpected personal information leaks, and nose generation by telephonic talks, etc. In this paper, the microphone system of the cellular phone is developed to prevent these problems caused by progress of information technology. The developed system was designed to detect only acoustic signals from a human being in the presence of various kinds of background noises. A windscreen was designed by use of micro-channels to eliminate the popping noise by the wind from the mouth of a speaker and four microphone array and signal processing techniques are applied to reduce background noise. The impact of the developed system was evaluated by experimental tests. The results show that the system can improve the required functions considerably.

• 한국정보통신학회논문지
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• 제21권4호
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• pp.708-717
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• 2017

음원 추적은 지능형 CCTV, 화상회의시스템, 음성 명령 인식 등에서 널리 활용되고 있다. 본 논문에서는 스마트강의시스템에 적합한 천정 부착형 직교배열 마이크로 음성 신호의 도착 시간차인 TDoA(Time Difference of Arrival)를 이용하는 실시간 음원추적 기법을 제안한다. TDoA를 위한 점 음원과 평행음원 모델을 분석하고, 3개의 선형배열마이크를 이용하여 상호 상관 방안을 제안하였다. 또한 직교축에 십자 배열된 5개의 마이크를 사용하여 전방위(omni-direction)에서 음원 추적이 가능함을 보였다. 무음구간을 제거하기 위하여 수신 에너지를 이용하였으며 상호상관을 부호로 연산하여 계산량을 줄이고 추정 결과에 미디언 필터(Median Filter)를 적용하여 안정도를 높였다. 제안된 시스템은 고속 MCU인 TMS320F379D와 MEMs마이크 모듈로 구현하여, 방향 검출 성능이 백색잡음이나 음악 등에 대해서는 0.5도, 음성에 대해서는 6.5도의 정밀도를 보였다.

• 한국수산과학회지
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• 제13권4호
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• pp.145-149
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• 1980

Various methods have been so far proposed to find out the directions and spectra of sound waves from the sources for provisions of noise controls. The conventional methods are generally classified into three systems such as, single microphone system, moving microphone system and multi-microphone system, which composes a resultant super directivity by giving a appropriate delay and a weighting coefficient in the output of each microphone. In case of using a single microphone there is a difficulty in providing it with desirable super directivity in the low frequency range, while in case of using multi-microphone system there has been a disadvantage that the measurement of directivity could not separately be done with the spectrum analysing. And in case of the use of a moving microphone system it needs a condition that the sound source to be detected should be stationary state and in rest. However here we introduce a method that the spectral analysing and the directivity of synthesis can be separately carried out by using a linear array of many microphones, in which each output of the microphone is multiplied by appropriate weighting coefficient and all of those products are summed after passing through adequate filters. The resultant signal is then sampled with an adequate sampling frequency and taken average for processing.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 추계학술대회 논문집
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• pp.225-228
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• 2009

The simulated indoor pass-by noise measurement system is the tool to measure and evaluate the pass-by noise at the test laboratory, without doing measurement at the field. This measurement system can realize the precision measurement under the specific condition and overcome the limitations of the field measurement, i.e. weather conditions, repeatability, .. This measurement system is done in time domain process using the array techniques, which synchronizes the time signals. The reliability of the obtained result depends on the array shapes, which can generate the moving source effect. In this paper, the validations are checked focusing the time domain synchronization of the signals with the optimum microphone array shape.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 1999년도 추계종합학술대회
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• pp.173-176
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• 1999

최근 화상 회의 시스템에서 화자 위치 추정 및 음질 향상 기술이 연구되고 있다. 이 시스템에서는 마이크로폰 어레이를 이용, 화자의 위치를 파악하여 화자의 방향으로 카메라를 자동으로 조정해 주게 된다. 본 연구에서는 마이크로폰 어레이를 통해 수신된 신호를 이용하여 Matched Filter Array를 구성하고 음질을 향상시켰다. 이때 역변환 필터로써 IIR필터를 사용하여 계산량을 줄였으며, 범용DSP 프로세서를 이용한 하드웨어를 제작하여 그 성능을 확인하였다.

• 한국인터넷방송통신학회논문지
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• 제14권1호
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• pp.31-36
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• 2014

실내에서의 위치기반 서비스를 위한 실내 위치 인식 기술에 대한 연구가 활발하다. 음향 신호와 TDOA 기술을 기반으로 하는 3차원 위치 인식 시스템에 있어서, 마이크 센서의 개수와 배열 위치에 따라 추정된 위치의 오차 특성이 달라지므로, 본 논문에서는 마이크 센서의 배열과 측정된 거리 차이값의 오차 크기에 따른 추정 위치의 오차를 DOP로 분석하고, 그 추정 위치 오차 패턴과 추정 위치 오차값을 고려하여 최적의 마이크 배열을 결정하는 방법에 대해서 연구하였다.

• 한국통신학회논문지
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• 제30권1C호
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• pp.54-60
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• 2005

본 논문은 지연 추정 기능을 갖는 적응 마이크로폰 어레이 알고리즘을 제안한다. Generalized sidelobe canceller (GSC)의 적응 차단 행렬이 각 센서간의 상호 시간 지연을 추정할 수 있다는 것을 보임으로써, 제안한 시스템은 적응 차단 행렬을 목적 신호의 차단 뿐 아니라 각 센서의 시간 지연 추정을 위해 사용한다. 이로 인해, 제안한 시스템은 GSC 구조만을 사용하면서. 시간 지연 추정기를 외부의 전처리기로 사용하는 기존의 시스템과 같은 성능을 얻을 수 있다. 실제 환경에서의 실험 결과를 통해 제안한 시스템의 성능이 기존의 시스템과 유사함을 확인하였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2005년도 춘계학술대회논문집
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• pp.164-169
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• 2005

Microphone array techniques are being used widely in wind tunnel measurements for identification of the distributed aerodynamic noise sources on the model being tested. Depending on the frequencies and sound levels, conventional beamforming algorithm has limitation in separating two adjacent sources. Several modifications to the classical beamforming have been developed to enhance way resolution and reduce sidelobe levels. In this Paper the robust adaptive beamforming and the CLEAN algorithm are used to compare to the result of conventional beamforming method. It is found that the CLEAN algorithm is capable of pin-pointing locations of multiple sources nearby, while these sources are unidentifiable with robust adaptive or conventional beamforming techniques.

• 한국군사과학기술학회지
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• 제21권4호
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• pp.447-455
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• 2018

If a hidden enemy is shooting, there is a threat against soldiers in recent conflicts. This paper aims to improve the localization of a muzzle using microphone array. Gunshot noise can provide information about the location of muzzle with two signals, the muzzle blast from the gun barrel and the projectile sound from the bullet. Two signals arrive to the microphone array with different arrival time and angle. If the arrival angles of the two signals are estimated, distance between sniper location and the microphone array can be calculated by using geometric principles. This method was established in 2003 by Pare. But this method has a limitation that it cannot calculate the distance when the arrival angles of the two signals are same. Also it has an error when the angle difference of arrival is small. In order to overcome this limitation, a new method is proposed that uses the change of characteristic of the projectile sound with respect to vertical distance from the trajectory. The proposed method estimates the distance correctly when the arrival angle of two signals are same, and when the angle difference between two signals is increased, the estimation error increases with respect to the angle. Therefore these two methods can be selected according to the angle difference between two signals to estimate the distance of the muzzle. Below the threshold of the angle difference, the proposed method can be used to estimate distance with smaller error than the existing method. This was demonstrated by shooting tests using actual sniper rifles.