• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.94-100
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• 1997

In order to investigate the radiation of sound from the King Song-Duk bell, we measured the sound pressure around the bell at every 30$^{\circ}$ using a microphone line array which was composed of 30 microphones separated by 15cm;total number of measurement point was 360. The sound field was estimated by using cylindrical acoustic holography. The spectrum of measured sound pressure demonstrates that it is almost like white noise in the very beginning, but in 10 seconds two close frequency components(64.06Hz, 64.38Hz) remain and make a famous beating. This beating sound is often believed to make unique sound of the King Song-Duk bell. The mode shapes of that frequency components are the same except that one is rotated by 45$^{\circ}$ from the other. This phenomenon occurs at the other pairs of components are the same except ones in very high frequency range where the mode shapes are rather complex.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.360-365
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• 2008

We propose an efficient robot name registration and recognition method in order to enable a Call-and-Come service for home robots. In the proposed method for the name registration, the search space is first restricted by using monophone-based acoustic models. Second, the registration of robot names is completed by using triphone-based acoustic models in the restricted search space. Next, the parameter for the utterance verification is calculated to reduce the acceptance rate of false calls. In addition, acoustic models are adapted by using a distance speech database to improve the performance of distance speech recognition, Moreover, the location of a user is estimated by using a microphone array. The experimental result on the registration and recognition of robot names shows that the word accuracy of speech recognition is 98.3%.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.36 no.3
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• pp.238-243
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• 2008

Several experimental techniques for identifying the noise sources distributed over a moving vehicle have been developed recently and are used to design a low noise vehicle. The beamforming method, which uses phase information between several microphones to localize the source position, is proved to be one of the promising techniques applicable even under complicated test environments. In this study a beamforming algorithm is developed and applied to measure the dominant noise sources on a passenger car passing by. Unlike the acoustic signals from a stationary noise source, the sound generated from a moving source is distorted due to the Doppler effects. The information about the speed and relative position of the vehicle are used to eliminate the Doppler effects from the measured acoustic signal by using a de-Dopplerization algorithm. The noise generated from a moving vehicle can be grouped in many ways, however, tire noise and the noise generated from the engine are distinguishable at the speeds being tested.

• Journal of IKEEE
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• v.23 no.4
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• pp.1360-1365
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• 2019

In this paper, we describe a sound source localization(SSL) system which can be applied to mobile robot and automatic control systems. Usually the SSL method finds the Interaural Time Difference, the Interaural Level Difference, and uses the geometrical principle of microphone array. But here we proposed another approach based on the deep neural network to obtain the horizontal directional angle(azimuth) of the sound source. We pick up the sound source signals from the two microphones attached symmetrically on both sides of the robot to imitate the human ears. Here, we use difference of spectral distributions of sounds obtained from two microphones to train the network. We train the network with the data obtained at the multiples of 10 degrees and test with several data obtained at the random degrees. The result shows quite promising validity of our approach.

• Advances in nano research
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• v.9 no.4
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• pp.263-276
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• 2020

In this article, the acoustic responses of free vibrated natural fibre-reinforced polymer nanocomposite structure have been investigated first time with the help of commercial package (ANSYS) using the multiphysical modelling approach. The sound relevant data of the polymeric structure is obtained by varying weight fractions of the natural nanofibre within the composite. Firstly, the structural frequencies are obtained through a simulation model prepared in ANSYS and solved through the static structural analysis module. Further, the corresponding sound data within a certain range of frequencies are evaluated by modelling the medium through the boundary element steps with adequate coupling between structure and fluid via LMS Virtual Lab. The simulation model validity has been established by comparing the frequency and sound responses with published results. In addition, sets of experimentation are carried out for the eigenvalue and the sound pressure level for different weight fractions of natural fibre and compared with own simulation data. The experimental frequencies are obtained using own impact type vibration analyzer and recorded through LABVIEW support software. Similarly, the noise data due to the harmonically excited vibrating plate are recorded through the available Array microphone (40 PH and serial no: 190569). The numerical results and subsequent experimental comparison are indicating the comprehensiveness of the presently derived simulation model. Finally, the effects of structural design parameters (thickness ratio, aspect ratio and boundary conditions) on the acoustic behaviour of the natural-fibre reinforced nanocomposite are computed using the present multiphysical model and highlighted the inferences.

• Journal of Institute of Control, Robotics and Systems
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• v.20 no.1
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• pp.48-57
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• 2014

It is suggested that the curvature trajectory be used to estimate the real-time moving sound sources and efficiently the robot estimating the sound sources. Since the target points of the real-time moving sound sources change, the mobile robot continuously estimates the changed target points. In such a case, the robot experiences a slip phenomenon due to the abnormal velocity and the changes of the navigating state. By selecting an appropriate curvature and navigating the robot gradually by using it, it is possible to enable the robot to reach the target points without having much trouble. In order to recognize the sound sources in real time, three microphones need to be organized in a straight form. Also, by applying the cross-correlation algorithm to the TDOA base, the signals can be analyzed. By using the analyzed data, the locations of the sound sources can be recognized. Based on such findings, the sound sources can be estimated. Even if the mobile robot is navigated by selecting the gradual curvature based on the changed target points, there could be errors caused by the inertia and the centrifugal force related to the velocity. As a result, it is possible to control the velocity of both wheels of the robot through the velocity PID controller in order to compensate for the slip phenomenon and minimize the estimated errors. In order to examine whether the suggested curvature trajectory is appropriate for estimating the sound sources, two mobile robots are arranged to carry out an actual experiment. The first robot is moved by discharging the sound sources, while the second robot recognizes and estimates the locations of the discharged sound sources in real time.

• The Journal of the Acoustical Society of Korea
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• v.11 no.3
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• pp.53-60
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• 1992

Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

• The KSFM Journal of Fluid Machinery
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• v.7 no.1 s.22
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• pp.17-23
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• 2004

Aeroacoustic characteristics of sirocco fan used in Over-The-Range (OTR) has been analyzed in this paper. A microwave hood combination over the gas range is short for the OTR. The flow phenomena of the double-sided sirocco fan was analyzed numerically and experimentally by using commercial code and three dimensional PIV for flow visualization. Also, microphone array is used in order to understand acoustic characteristics of OTR. Two dimensional unsteady flow and acoustic simulation is tried to qualitatively estimate the effects of tonal noise and broadband noise on the overall sound pressure level. It is found that tonal sound is generated from the strong interaction between the impeller and cutoff while broadband sound is generated from the strong secondary flows along the scroll surface. To reduce the noise level, the V-shape cut-off was applied to improve the sound quality by reducing tonal noise. So the peak noise at BPF (Blade Passing Frequency) was almost reduced. The shape of flow-guide to suppress the secondary flow over the scroll surface was carefully checked. It is found that this affects flow pattern at the fan exit and reduces the broad band noise. Through this numerical and experimental study, the sound pressure level was lowered by 4dBA compared to that of the previous fan at the operating point.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.102-109
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• 2016

This paper proposes a noise suppression method through SNR (Singal-to Noise Ratio) estimation in the two microphone array environment of close spacing. The conventional method uses a noise suppression method for a gain function obtained through the SNR estimation based on coherence function from full band. However, this method cause performance decreased by the noise damage that affects all the feature vector component. So, we propose a noise suppression method that allocates a frequency domain signal into N constant multi frequency band and each frequency band gets a gain function through SNR estimation based on coherence function. Performance evaluation of the proposed method is shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation method provided by the ITU-T (International Telecommunications Union Telecommunication).

• The Journal of the Acoustical Society of Korea
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• v.38 no.4
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• pp.406-413
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• 2019

This paper proposes a speech enhancement method through the process of combining the gain function with spectrum subtraction method in the two microphone array with close spacing. A speech enhancement method that uses a gain function estimated by the SNR (Signal-to Noise Ratio) based on the multi frequency band coherence function causes the performance degradation in high correlation between input noises of two channels. A new speech enhancement method is proposed where the weighted gain function is used by combining the gain function from the spectral subtraction. The performance evaluation of the proposed method was shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation test provided by the ITU-T (International Telecommunications Union Telecommunication). In the PESQ tests, the maximum 0.217 of PESQ value is improved in the various background noise environments.