• Journal of IKEEE
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• v.14 no.1
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• pp.47-54
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• 2010

In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.

• ETRI Journal
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• v.27 no.2
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• pp.153-165
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• 2005

The main purpose of a spatial audio system is to give a listener the same impression as if he/she were present in a recorded environment. A dummy head microphone is generally used for such purposes. Because of its human-like shape, we can obtain good spatial sound images. However, its shape is a restriction on its public use and it is difficult to convert a 2-channel recording into multi-channel signals for an efficient rendering over a multi-speaker arrangement. In order to solve the problems mentioned above, a spatial audio system is proposed that uses multiple microphones on a rigid sphere. The system has five microphones placed on special points of the rigid sphere, and it generates audio signals for headphone, stereo, stereo dipole, 4-channel, and 5-channel reproduction environments. Subjective localization experiments show that front/back confusion, which is a common limitation of spatial audio systems using the dummy head microphone, can be reduced dramatically in 4-channel and 5-channel reproduction environments and can be reduced slightly in a headphone reproduction.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.520-524
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• 2002

Multi-reference, scan-based Acoustical Holography is a useful measurement technique when insufficient microphones are available to measure a complete hologram at once. When the sound sources are stationary, the whole hologram can be constructed by joining together sub-holograms captured using a relatively small scan array. Here that approach is extended by the development of a formulation that explicitly includes the acoustical transfer functions between the reference microphones and the scanning microphones. Based on those expressions, a compensation procedure of spectral variance due to source-non-stationarity is proposed. It has been verified both numerically and experimentally that this procedure can help suppress spatially distributed noise caused by the source level non-stationarity that is always present in a measurement.

• Journal of Broadcast Engineering
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• v.23 no.4
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• pp.519-525
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• 2018

This paper deals with noise suppression of multi-channel data captured by microphone array using multi-channel Wiener filter. Multi-channel Wiener filter does not rely on information about the direction of the target speech and can be partitioned into an MVDR (Minimum Variance Distortionless Response) spatial filter and a single channel spectral filter. The acoustic transfer function between the single speech source and microphones can be estimated by subspace decomposition of multi-channel Wiener filter. The errors are incurred in the estimation of the acoustic transfer function due to the errors in the estimation of correlation matrices, which in turn results in speech distortion in the MVDR filter. To alleviate the speech distortion in the MVDR filter, diagonal loading is applied. In the experiments, database with seven microphones was used and MFCC distance was measured to demonstrate the effectiveness of the diagonal loading.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.596-600
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• 2009

We are developing a simple low-cost wind velocity sensor based on small microphones. The sensor system consists of 4 microphones covered with specially shaped wind screens, 4 pre-amplifiers that respond to low frequency, and a commercial sound interface with multi channel inputs. In this paper, we first present the principle of the sensor, i.e., technique to successfully suppress the influence of external noise existing in the environment in order to determine the wind velocity and the wind direction from the output from a microphone. Then, we present an application for generating realistic motions of a virtual tree swaying in real wind. Although the current sensor outputs significant leaps in a measured sequence of directions, the interactive animations demonstrate that it is usable for such applications, if we could reduce the leaps to some degree.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05b
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• pp.96-102
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• 2003

Today, stage technology is developing highly by application of digital computer. Performance is composed of audio/video and acoustic technology takes very important position in field of stage technology. Generally speaking, four factors of sound are loudness, pitch, sound timbre and duration. Loudness depends on sound pressure level, yet partly related with spectrum and dulation. Pitch depends mainly on frequence and have a relation with sound pressure and duration. sound timbre depends strongly on spectrum and have a relation with frequence. In this paper, I designed a multi-microphone system which can used in broadcasting and performance stage with vicboss 200MHz-VHF wireless microphone and vicboss 900MHz-VHF wireless microphone. I also studied about multi-microphone which can use conveniently in the super play that needs many microphones. If this multi-microphone is prodused, we could expect better sound quality and a big progress in stereo recording technology.

• Journal of KSNVE
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• v.9 no.6
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• pp.1137-1144
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• 1999

Most of ANC(active noise control) researches are focused on adaptive algorithms, computer simulations and implementations of single-channel system in experimental environments. In this paper, a multi-channel ANC system based on DSP's was developed to obtain global attenuations over wide region and applied to the active control of the humming noise generated by a transformer. The developed ANC system including 24 microphones and 12 spearkers was applied to the real transformer noise reduction problem. Results showed that the control system could successfully control the humming noise over the region of interest.

• Journal of Korea Multimedia Society
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• v.18 no.3
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• pp.323-329
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• 2015

In this paper we applied pattern recognition approach to detect audio forgery. Classification of the microphone types and models can help determining the authenticity of the recordings. Canonical correlation analysis was applied to extract feature for microphone classification. We utilized the linear dependence between two near-silence regions. To utilize the advantage of multi-feature based canonical correlation analysis, we selected three commonly used features to capture the temporal and spectral characteristics. Using three different microphones, we tested the usefulness of multi-feature based characteristics of canonical correlation analysis and compared the results with single feature based method. The performance of classification rate was carried out using the backpropagation neural network. Experimental results show the promise of canonical correlation features for microphone classification.

• The Journal of Korea Robotics Society
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• v.16 no.4
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• pp.291-298
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• 2021

In this study, we present a multi-sensor module for snake robot searching survivors in a narrow space. To this end, we integrated five sensor systems by considering the opinions of the first responders: a gas sensor to detect CO₂ gases from the exhalation of survivors, a CMOS camera to provide the image of survivors, an IR camera to see in the dark & smoky environment, two microphones to detect the voice of survivors, and an IMU to recognize the approximate location and direction of the robot and survivors. Furthermore, we integrated a speaker into the sensor module system to provide a communication channel between the first responders and survivors. To integrated all these mechatronics systems in a small, compact snake head, we optimized the positions of the sensors and designed a stacked structure for the whole system. We also developed a user-friendly GUI to show the information from the proposed sensor systems visually. Experimental results verified the searching function of the proposed sensor module system.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.4
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• pp.355-362
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• 2009

Sound source localization method based on the time delay of arrival(TDOA) is applied to many research fields such as a robot auditory system, teleconferencing and so on. When multi-microphones are utilized to localize the source in 3 dimensional space, the conventional localization methods based on TDOA decide the actual source position using the TDOAs from all microphone arrays and the detection measure, which represents the errors between the actual source position and the estimated ones. Performance of these methods usually depends on the number of microphones because it determines the resolution of an estimated position. In this paper, we proposed the localization method using spatially mapped GCC functions. The proposed method does not use just TDOA for localization such as previous ones but it uses spatially mapped GCC functions which is the cross correlation function mapped by an appropriate mapping function over the spatial coordinate. A number of the spatially mapped GCC functions are summed to a single function over the global coordinate and then the actual source position is determined based on the summed GCC function. Performance of the proposed method for the noise effect and estimation resolution is verified with the real environmental experiment. The mean value of estimation error of the proposed method is much smaller than the one based on the conventional ones and the percentage of correct estimation is improved by 30% when the error bound is ${\pm}20^{\circ}$.