• 전기학회논문지
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• 제57권12호
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• pp.2300-2307
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• 2008

This study receives the noise transmitted in a constant audio frequency range through a microphone array in which the noise(like grease in a pan) occurs on the power supply line due to the troublesome partial discharge(arc). Then by going through a series of signal processing of removing noise, this study measures the distance and direction up to the noise caused by the troublesome partial discharge(arc) and monitors the result by displaying in the analog and digital method. After these, it determines the state of each size and judges the distance and direction of problematic part. When the signal sound transmitted by the signal source of bad insulator is received on each microphone, the signal comes only in the frequency range of 20 kHz by passing through the circuit of amplification and 6th low pass filter. Then, this signal is entered in a digital value of digital signal processing(TMS320F2812) through the 16-bit A/D conversion. By doing so, the sound distance, direction and coordinate of bad insulator can be detected by realizing the correlation method of detecting the arriving time difference occurring on each microphone and the algorithm of detecting maximum time difference.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1999년도 제14차 학술회의논문집
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• pp.266-269
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• 1999

An electronically Scannable microphone system is in the Planning stage. For this Purpose, a multiple microphone array with controllable delay is available. To achieve effective point-listening characteristics, we proposed an enclosed microphone array system with a complex weighting method. In this system, both the microphone arrangement and the value of the complex weighting are important. In this report, the construction of microphone array system and the signal-processing method are explained, and the calculation method for optimal complex weighting is also presented. A prototype experimental setup is designed and fabricated to verify the expected characteristics.

• The Journal of the Acoustical Society of Korea
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• 제17권2E호
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• 한국음향학회지
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• 제30권3호
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• pp.115-122
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• 2011

본 논문에서는 십자 형태로 구성된 네 개의 마이크로폰을 이용하여 일반화된 상호 상관 기법을 적용한 음성 도달시간 지연을 측정하여 정확한 음원 방향을 실시간으로 계산하는 방식에 대해 제시하였다. 기존 시스템에서는 마이크로폰 어레이 신호처리를 위해 데이터 수집 장치를 필요로 하므로 시스템을 내장하기 힘들고, 또한 DSP 프로세서를 사용한 음원방향 인지는 마이크로폰의 채널의 수가 늘어날수록 실시간 처리가 어려워지는 두 가지 단점이 있다. 본 논문에서는 이러한 한계를 극복하기 위하여 마이크로폰 어레이 신호처리를 이용한 향상된 음원방향 인지 하드웨어의 개발을 제안하였다. 공간 구분 기법을 이용한 효율적인 설계 및 검증방식을 제안하였고 이를 통하여 보다 정확한 방향 추정과 설계시간 단축이 가능하다. 최종적으로 음성 코덱과 FPGA를 이용하는 임베디드 시스템을 위해서 사용이 가능한 시스템을 개발하였다. 실험 결과에 의하면 PC 기반이나 DSP 프로세서를 사용한 경우에 비해 보다 빠른 처리 시간을 보였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2011년도 춘계학술대회 논문집
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• pp.816-820
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• 2011

본 논문은 구형 마이크로폰 어레이의 부엽 레벨의 차를 증가시키기 위한 방법에 대한 연구 내용을 다루었다. 일반적인 어레이 신호처리에서 마이크로폰을 조밀하게 배치함으로써 어레이 응답에서의 주엽과 부엽 간의 차이를 늘릴 수 있고 어레이의 소음원 판별능력을 증가시킨다. 최근 사용되고 있는 상용 에레이들은 제작 단가와 어레이의 크기 때문에 센서의 수를 늘리는데 한계를 보이고 있다. 이런 문제를 극복하기 위해 본 연구에서는 MEMS 센서를 이용하여 구형 어레이에 적용하였다. 구형 마이크로폰 어레이를 이용한 시뮬레이션과 실험을 통해 정현파 소음원을 측정하였다. 실험을 위해 32 개의 일반 측정용 마이크로폰을 이용한 어레이와 85 개의 MEMS 마이크로폰을 이용한 구형 어레이를 제작하였다. 구형 조화 분해기법과 빔형성기법을 이용하여 측정 데이터를 분석하였다. 2 kHz 이상의 소음원에 대하여 MEMS 마이크로폰 어레이가 4 dB 이상의 부엽 저감 능력을 가지는 것을 확인하였다.

• 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.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 1998년도 제13차 학술회의논문집
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• pp.75-80
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• 1998

This paper proposes a novel computer human interface, named Virtual Wireless Microphone (VWM), which utilizes computer vision and signal processing. It integrates real-time face tracking and sound signal processing. VWM is intended to be used as a speech signal input method for human computer interaction, especially for autonomous intelligent agent that interacts with humans like as digital secretary. Utilizing VWM, the agent can clearly listen human master's voice remotely as if a wireless microphone was put just in front of the master.

• 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.

• 한국통신학회논문지
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• 제29권11C호
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• pp.1573-1580
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• 2004

본 논문은 실제 환경에서 고품질 음성 신호 취득을 위해, 적응 마이크로폰 어레이 시스템을 위한 적응 모드 컨트롤러를 제안한다. 적응 어레이 알고리즘의 올바른 동작을 위하여, 제안된 적응 모드 컨트롤러는 시간 축의 정보뿐만 아니라 공간 축의 정보를 함께 사용한다. 제안된 적응 모드 컨트롤러는 초기화 단계와 수행 단계의 두 단계로 나뉘어 동작되는데, 초기화 단계에서는 음원 위치 추정 기술이 사용되며, 수행 단계는 신호의 상관 관계 특성에 의해 동작한다. 적응 어레이 알고리즘으로는 적응 차단 행렬을 이용한 Generalized Sidelobe Canceller가 사용되었다. 제안한 적응 모드 컨트롤러는 적응 차단 행렬이 수렴되지 않은 경우에도 사용 가능하며, 기존의 전력비 방법에 비해 안정적인 성능을 나타낸다. 본 논문은 제안한 시스템을 실제 환경에서 평가하였으며, 2m 거리에 위치한 화자에 대해 13dB SINR 향상을 얻었다.

• 대한음성학회지:말소리
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• 제66호
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• pp.73-86
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• 2008

Beamforming is one of the spatial filtering techniques which extract only desired signals from noisy environments using microphone arrays. Fixed beamforming is a simple concept and easy to implement. However, it does not show good performance in real noisy conditions. As an adaptive beamforming, Frost algorithm can be a good candidate. It uses the concept of the linearly constrained minimum variance (LCMV) algorithm. The difference between the Frost and the LCMV algorithm is the error correction scheme which is very effective feature in the aspect of performance. In this paper, as quadrature mirror filtering (QMF)-based filterbank is utilized as the pre-processing of the Frost beamformning, the filter length and the learning rate of each band is optimized to improve the performance. The performance is measured by the signal-to-noise ratio (SNR) and the Bark's scale spectral distortion (BSD).