• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.05a
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• pp.1003-1008
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• 2001

In beamforming method, source positions are predicted by MUSIC (Multiple Signal Classification) power method and composite sound fields can then be decomposed into each partial field by beamforming, detenninistically without restriction of the distance between reference microphones and sources. However, reference microphone array shape is important in both MUSIC and beamforming method. Thus the present paper describes the effect of the reference microphone array shape.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.4
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• pp.624-633
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• 2018

While the multimedia lecture system has been getting smart using immerging technology, the microphone still relies on the classical approach such as holding in hand or attaching on the body. In this paper, we propose a ceiling mounted array microphone system that allows a wide reception coverage and instructors to move freely without attaching microphone. The proposed system adopts cell and handover of mobile communication instead of a complicated beamforming method and implements a wide range microphone over several cells with low cost. Since the characteristics of unvoiced speech is similar to Pseudo Noise it is shown that soft handover are possible with 3 microphones connected to delay-sum multipath receiver. The proposed system is tested in $6.3{\times}1.5m$ area. For real-time processing the correlation range can be reduced by 82% or more, and the output latency delay can be improved by using the delay adaptive filter.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.92-96
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• 2007

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet windtunnel.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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• pp.484-488
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• 2007

In this paper, the measurement system using the microphone array developed for analyzing cabin noise of the vehicle and its applications are discussed. The sensor is a three dimensional microphone array, the microphones and cameras are equipped on the rigid sphere. The cameras are used for acoustic visualization. As applications, the experiments in both reverberation chamber and anechoic chamber are discussed. These results show that this system is very useful to evaluate or improve the vehicle sound insulation performance.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• Journal of KSNVE
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• v.6 no.6
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• pp.835-842
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• 1996

To identify the locations and strengths of acoustic sources, one may use a microphone line array. Apparent advantage of the source identification method utilizing a line array is that it requires less measurement points than intensity method and holography. This method is based on the information of magnitude and phase difference between pressure signals at each microphone. Since those differences are dependent on the source model, we have to assume them such as plane, monopole, etc. In this paper the conventional source identification methods such as beamforming method and MUSIC method are briefly reviewed by modeling a source as plane and spherical wave, then a modified method is introduced. This can be applied to sound field which may by either coherent or incoherent. Typical simulations and experiment are performed to confirm this identification method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.6
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• pp.612-618
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• 2008

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet wind tunnel.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.701-708
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• 2003

This paper presents an acoustic source localization technique on 2D cavity model in flow using a phased microphone way. Investigation was performed on cavity flows of open and closed types. The source distributions on 2D cavity flow were investigated in anechoic open-jet wind tunnel. The array of microphones was placed outside the flow to measure the far field acoustic signals. The optimum sensor placement was decided by varying the relative location of the microphones to improve the spatial resolution. Pressure transducers were flush-mounted on the cavity surface to measure the near-filed pressures. It is shown that the propagated far field acoustic pressures are closely correlated to the near-field pressures. It is also shown that their spectral contents are affected by the cavity parameter L/D.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1E
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• pp.45-53
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• 2010

In this paper, we present the use of the Features Weighted Mahalanobis Distance (FWMD) in improving the performance of Likelihood Maximizing Beamforming (Limabeam) algorithm in speech recognition for microphone array. The proposed approach is based on the replacement of the traditional distance measure in a Gaussian classifier with adding weight for different features in the Mahalanobis distance according to their distances after the variance normalization. By using Features Weighted Mahalanobis Distance for Limabeam algorithm (FWMD-Limabeam), we obtained correct word recognition rate of 90.26% for calibrate Limabeam and 87.23% for unsupervised Limabeam, resulting in a higher rate of 3% and 6% respectively than those produced by the original Limabearn. By implementing a HM-Net speech recognition strategy alternatively, we could save memory and reduce computation complexity.