• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.22-27
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• 1998

A modified perturbation method is proposed to optimize the beam pattern of thinned microphone arrays. Both microphone spacing and array weight are iteratively adjusted via successive perturbation to achieve an optimum beam pattern in a Dolph Chebyshev sense. To improve the sidelobe performance, an alternative perturbation with respect to microphone spacing and array weight is implemented. Also, a linear space-tapering is employed in the perturbation process. It is demonstrated that the proposed approaches successfully yield sidelobe performances comparable to that of a normal array. Computer simulation results are presented.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.38-45
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• 2005

The noise reduction of the In-The-Ear (ITE) hearing aid (HA) can be achieved by arrays of microphones. Each of the right and the left ears was assumed to have two HA microphones. These arrays of HA microphones produce particular patterns of directivity by some time delay between two microphones. The directivity pattern geometrically increase the S/N ratio. The boundary element method (BEM) was used for the three dimensional simulation of the HA directivity pattern with the two pairs' microphone arrays. The separation between two microphones was fixed to 10 mm. The time delay between the two microphones was calculated to produce the most narrow directivity pattern in the fore front of the head. The variation of the time delay was examined in accordance with input frequencies. This numerical analysis may be then applied for the calculation of the time delay parameter of the digital hearing aid DSP chip.

• Proceedings of the KSR Conference
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• 2011.10a
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• pp.2269-2275
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• 2011

This paper deals with the design of a microphone array to measure location and spectral characteristics of railway noise sources. A microphone array estimates the direction of a noise source assumed as a point source using the delayed-sum beamforming method and its performance is determined in terms of resolution and side-lobe level. A 48 channel microphone array was already developed to measure noise sources of KTX trains and a new array with 96 microphones has been designed to enhance the performance. This paper simulates the performance the microphone arrays according to the configuration of microphones and verifies it through on-site tests.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.10a
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• pp.463-464
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• 2009

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.05a
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• pp.742-743
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• 2010

• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.816-820
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• 2011

A method for increasing the difference of side-lobe level in spherical microphone array is presented. In array signal processing, it is known that narrow interval between sensors can increase the difference between main lobe and side-lobe of array response which eventually increase the source recognition capability. Recent commercial array being used, however, have shown certain limitation in using the number of sensors due to its costs and geometrical size of array. To overcome this problem, we have adapted MEMS sensors into spherical microphone array. To check out the improvement, two different types of spherical microphone array were designed. One array is composed with 32 regular instrument microphones and the other one is 85 MEMS sensors. Simulation and experiments were conducted on a sinusoidal noise source with two arrays. The time history data were analyzed with spherical harmonic decomposition and beamforming technique. 85 MEMS sensors array showed the improved side-lobe level suppression by more than 4 dB above the frequency content of 2 kHz compared to 32-sensor array.

• 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).

• Proceedings of the KSR Conference
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• 2006.11b
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• pp.144-151
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• 2006

Sound generated from a moving vehicle often carries information on the condition of vehicle, for example, whether it has faults or not, where the fault exists. The latter is possible especially by MFAH(moving frame acoustic holography) and beamforming method. MFAH is applicable to the sound source of pure tone or narrow band noise. For the beamforming method, we have to know what kind of wave the sound source radiates, for example, plane wave or spherical wave. That is, whether the above methods are applicable depends on the characteristics of sound source. To apply these methods to the fault detection, we have to know the characteristics of wave from faults. In this research, a machine diagnosis technique based on the above holographic approaches is introduced to find the position of faults. The signal due to faults is modeled based on the fact that the faults radiate impulsive noise, and analyzed in time and frequency domain. The way how MFAH and beamforming method can be used is introduced to find the position of source.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.446-449
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• 2003

The conventional GSC is inappropriate in real situation when the target signal is present. The steering vector error cancels the target signal and the target signal misadjusts the weight of the adaptive filter. To prevent the target signal cancellation, the robust GSC using the constrained adaptive filters was already proposed. However, the adaptive weight misadjustment is not settled in robust GSC. This Paper proposes a revised robust sidelobe canceller with adaptive compensator. To compensate the influence of target signal, the adaptive compensator is used in cascade. In computer simulation, we show the performance improvement by comparing the robust GSC with the proposed GSC.