• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.158-163
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• 2005

In this paper the aeroacoustic characteristics of a tilt-rotor system is measured by using a sealed model tilt-rotor. With a microphone array system and the do-dopplerization algorithm, the location and the characteristics of rotor noise are successfully measured. The most of high frequency noise (4kHz) is found to be located at rotor blade tips, but the low frequency tonal noise is dominant on the middle of the rotor blades. The measured tonal noise characteristics are compared to the results of theoretical calculation. At 0.5m distance from the rotor plane, measured and calculated data are relatively well matched regardless of rotating speed and collective pitch angie for the azimuthal angles between $0^{\circ}\;and\;60^{\circ}$. However, the data on the azimuthal angles between $70^{\circ}\;and\;90^{\circ}$ are not quite comparable. In addition, the compared data for far-field case (1.5m) are quite different. This is probably due to the unsteady effect which if not included in the theoretical calculation.

• Journal of the Korean Society for Railway
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• v.15 no.1
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• pp.23-28
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• 2012

In this paper, noise of Korean high-speed trains (KTX) running at different speed from 150 to 300km/h was measured by a microphone array system. From the measurement, relation between maximum sound pressure levels and train moving speeds of KTX was drawn and a regression coefficient from the relation was also derived. Moreover, increases of SPL with speeds of KTX were analyzed in the frequency domain. From the analysis, sound characteristics of passing-by noise of KTX were provided. Then, dominant noise source areas were obtained from the measurements and propagation patterns of KTX in vertical direction were also investigated. Finally, noise sources of KTX were identified from inspection of noise maps.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.30-35
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• 2006

In this paper. we Propose a frequency domain beamforming algorithm composed of inverse-filtering stages followed by a MVDR (Minimum-Variance Distortionless Response) beamformer or a GSC (Generalized Sidelobe Canceller). The proposed method is shown to require less complexity than the conventional RTF-MVDR and TF-GSC. respectively, and it is shown that the Proposed method is equivalent to the conventional RTF-MVDR and TF-GSC in optimum solution. In order to evaluate the performance of the Proposed method. speech recognition experiments are performed using the speech database recorded in a car. The Proposed method shows equal or slightly degraded Performance comparing to the conventional methods in terms of the speech recognition rate.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.358-367
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• 2016

SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

• The Journal of the Acoustical Society of Korea
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• v.34 no.1
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• pp.60-65
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• 2015

In this paper, we used the sound of gunshots recorded by multiple microphones to increase the accuracy of the calculation of the distance between sniper and the microphone array. This method is crucial for achieving military objectives. Gunshots are comprised of the explosion of driving gas from the muzzle and the supersonic shock wave from the flying bullet. The original distance calculation method compares the time difference of arrival and angle of incidence to estimate the sniper's location. The disadvantage of this method is that when the angles of incidence coincide the margin of error increases, to solve this problem we suggest a new method using the characteristic changes of the shock wave with the increase of perpendicular distance between the microphone and the trajectory of the bullet. This theory is verified by experiments.

• 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 the Institute of Electronics and Information Engineers
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• v.51 no.10
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• pp.256-263
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• 2014

We have designed and made three kinds of FBG(Fiber Bragg Grating) Acoustic Transducer using Hopper type WDM on the use of recently developed FBG in Korea. The newly designed three kinds of FBG Acoustic Transducer using Hopper type WDM have an excellent merit of practical use with simple structure of sensors arm as well as the merit with existing fiber sensors. It was possible to detect sound waves in the range of 10 Hz to 18 kHz through the newly designed three kinds of FBG Acoustic Transducer and also, possible to detect its signal within the maximum range of 8.6 m by the use of most suitable resonance condition of the transducer. Especially, we can expect the utilization of low-frequency signal detection instead of existing acoustic sensor in the environment of electric noise and inferior condition. Furthermore, they can be developed as the high-sensibility and multi-point signal detection system through the sensor array system.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.102-105
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• 2000

This paper propose new sound localization algorithm that calculates TDOA(Time Difference Of Arrival) between the two received signals via two microphone array, The proposed Subband CPSP is a development of Previous CPSP method using subband approach. It first split the received microphone signals into three frequency bands and then calculates subband CPSP with corresponding SNR weights. This type of algorithm, Subband CPSP, can provide more accurate TDOA estimation results because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, computer simulation was conducted and it was compared with previous CPSP method. From the both simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than accuracy for TDOA estimation.

• MALSORI
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• no.66
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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).

• The Journal of the Acoustical Society of Korea
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• v.25 no.1E
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• pp.25-31
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• 2006

Two twin microphones may produce particular patterns of binaural directivity by time delays between the twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the KEMAR head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern. The simulation results were verified by the experimental measurement.