• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.334-337
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• 2005

The speech intelligibility is one of the most important factors in communication system. The speech intelligibility is related with speech to noise ratio. To enhance the speech to noise ratio, background noise reduction techniques are being developed. As a part of solution to noise reduction, this paper introduces directional microphone using beamforming method and speech filtering method. The directional microphone narrows the spatial range of processing signal into the direction of the target speech signal. The noise signal located in the same direction with speech still remains in the processing signal. To sort this mixed signal into speech and noise, as a following step, a speech-filtering method is applied to pick up only the speech signal from the processed signal. The speech filtering method is based on the characteristics of speech signal itself. The combined directional microphone and speech filtering method gives enhanced performance to speech intelligibility in communication system.

• ETRI Journal
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• v.36 no.5
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• pp.772-782
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• 2014

Two-microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time-frequency (T-F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T-F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T-F units with hit rates above 85%. It outperforms previous solutions in terms of signal-to-noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.

• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• Journal of KSNVE
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• v.4 no.2
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• pp.187-198
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• 1994

This study is to describe the effects of the duct termination conditions conditions upon the active noise attenuation system. The adaptive filtering algorithm using FIR filter is implemented for duct noise attenuation. To avoid the instability caused by the acoustic feedback, two methods are considered. One is to use a compensating FIR filter. The other is to utilize uni-directional detecting microphone and uni-directional control speaker. Experimental results show that the reflections of sound from duct terminations greatly reduce the performance of ANC system. The directionality of detecting microphone and control speaker is a major factor to decide ANC performance. When there are some reflections from both duct terminations, the noise attenuation using finite FIR filter is not enough to model the duct plant. Especially, the reflection from the upstream termination reduces the noise attenuation in the frequencies related to the distance between control speaker and upstream termination. The performance of the noise attenuation is found to be largely enhanced by using uni-directional coupler, both on detecting microphone and control speaker, even if the duct system has an arbitrary termination conditions.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.11a
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• pp.735-741
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• 2003

A simplified view of array design and application process was introduced. Array design is critical to achieve a successful phased array measurements. A planar microphone array is designed to produce optimum performance and also to fit economic requirement in integrating data acquisition system. Certain performance characteristics are of primary concern when designing arrays. These characteristics include array resolution, spatial aliasing and array sidelobe suppression. Every array has its directional pattern that shows such characteristics. Assuming that a monopole source is located in center, beam-patterns have been simulated varying measurement conditions such as number of sensors. array aperture size, distance between array and source, frequency of interest and so on. Sensor correction was conducted on very channel using magnitudes and phased of FRF with respect to a reference microphone channel. Then with a spiral type array, measurements have been made with two point sources of same frequency in order to investigate array resolving abilities. It is observed that higher frequency source achieves better resolution than lower one does.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.197-204
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• 2012

In this thesis, studied the method to minimize the changes in frequency response and level due to the variation of the distance from the source to the microphone. selecting three microphones (omni directional, cardioid, super cardioid) which are being used generally, frequency responses were measured in accordance with the distance changes. Gotten the difference from the reference as the result of measurement, changed responses for each frequency range were compensated in comparison of the original human vocal source. In low frequency range, the low frequency boost caused by the proximity effect and decrease in accordance with the distance were compensated. The variation in mid-frequency range is comparatively small, however since the mid-range is the most important part of the human vocal signal, were compensated the mid-frequency range in comparison of the reference. The human vocal signal variation in high frequency range is extremely small and the high frequency is compensated close to the original source without difficulty. Understanding the microphone characteristics and compensations, this study showed that the response can be maintain among the change of the distance from the source to the microphone.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.47-53
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• 1997

This paper concerns an efficient approach to forming a beam pattern of a microphone array to deal with broadband signals such as speech in a teleconference. A numerical method is proposed to find updated location of sidelobes for equalizaing the sidelobes via perturbation of array parameters such as array weight or microphone spacing. Thus the microphone array is optimized in a Dolph-Chebyshev sense such that directional or background noises incident in an array visual range are eliminated efficiently. It is shown that perturbation of microphone spacing yields an optimum pattern more appropriate for dealing with broadband signals than that of array weight. Also, a novel method is proposed to find a beam pattern which is robust with respect to sidelobe in a scanning situation. Computer simulation results are presented.