• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.620-628
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• 2005

Spatial control of sound is essential to deliver better sound to the listener's position in space. As it can be experienced in many listening environments, the quality of sound can not be manifested over every position in a hall. This motivates us to control sound in a region we select. The primary focus of the developed method has to do with the brightness and contrast of acoustic image in space. In particular, the acoustic brightness control seeks a way to increase loudness of sound over a chosen area, and the contrast control aims to enhance loudness difference between two neighboring regions. This enables us to make two different kinds of zone - the zone of quiet and the zone of loud sound - at the same time. The other perspective of this study is on the direction of sound. It is shown that we can control the direction of perceived sound source by focusing acoustic energy in wavenumber domain. To begin with, the proposed approaches are formulated for pure-tone case. Then the control methods are extended to a more general case, where the excitation signal has broadband spectrum. In order to control the broadband signal in time domain, an inverse filter design problem is defined and solved in frequency domain. Numerical and experimental results obtained in various conditions certainly validate that the acoustic brightness, acoustic contrast, direction of wave front can be manipulated for some finite region in space and time.

• Journal of Mechanical Science and Technology
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• v.15 no.2
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• pp.232-238
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• 2001

Fast Fourier Transformation is employed to convert the head variation of a pipeline in the time domain to the amplitude of the frequency domain. Applying method of characteristics to a pipeline provides a significant frequency range for a surge introduced from the valve modulation. Inverse Fast Fourier Transformation and a Finite Impulse Response Filter can be used to remove any possible noise existing from the significant frequency range of an unsteady condition. A filtered signal shows higher potential for the inverse calculation of leakage detection than the noise-added signal does. The respective performances of Inverse Fast Fourier Transformation and a Finite Impulse Response Filter are compared in terms of leakage detection capability. Characteristics of the frequency range for multiple leakages were investigated to validate the effectiveness of the noise control method in the frequency domain.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.225-228
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• 2000

It is well known that a Frequency Domain Adaptive Filter (FDAF) converges faster than a Time Domain Adaptive Filter (TDAF) even when the input signal is colored such as a speech signal. We have proposed the FDAF using the Modified Discrete Fourier Transform Pair (MDFTP) and its realization and effectiveness has been confirmed through the computer simulations. In this paper, we apply the FDAF using the MDFTP to the noise cancellation system. The proposed system is based on the Adaptive Line Enhancer (ALE) and utilizes single microphone; therefore it is suitable for the portable electronic equipment. Moreover, we propose to utilize the MDFT for detecting of the pitch in the speech because the number of data points in the MDFT must be equal to the pitch to confirmed that the noise can be removed to near the level of SNR.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.7
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• pp.569-574
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• 2014

In order to investigate low frequency swishing noise of wind turbines, acoustic source model using a singularity in circular motion is introduced to derive analytic solution of Lowson acoustic analogy in time domain. Results in time and frequency domains computed by the solution show apparent modulation of amplitude and frequency. The solution indicates that time histories of acoustic pressure at receiver points varied significantly according to receiver's directional location, even when the retarded time distributions are similar. However, the corresponding time-averaged spectra of sound pressure at the receiver locations where the retarded time distributions are almost same are not significantly different. It can be inferred from these results that the time-averaged sound pressure spectra which cannot take into account the detailed difference in the time-variation of wind turbine noise may not represent the sound quality of wind turbines due to its swishing. Finally, as an introduction of procedure to quantify low frequency swishing noise level, relative variation of overall sound pressure level is obtained using tonal low frequency noise model.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.837-842
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• 2000

Reverberation time is the well known theory and widely used in commercial apparatus to get reverberation time. However large fluctuation in low frequency region occurs in a small cabin due to superposition of a few modes. This paper investigates this phenomena in terms of modal density in frequency domain and suggests a method to get lower limit of reverberation time using the integration of the time-SPL diagram. The suggestion is confirmed by simulation and shows reasonable results to get lower limit of reverberation time and maximum absorbing power in the cabin.

• Journal of the Optical Society of Korea
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• v.15 no.1
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• pp.103-108
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• 2011

Recently reported asynchronous-optical-sampling terahertz (THz) time-domain spectroscopy enables high-resolution spectroscopy due to a long time-delay window. However, a long-lasting tail signal following the main pulse is often measured in a time-domain waveform, resulting in spectral fluctuation above a background noise level on a high-resolution THz amplitude spectrum. Here, we adopt the wavelet power spectrum estimation technique (WPSET) to effectively remove the spectral fluctuation without sacrificing spectral features. Effectiveness of the WPSET is verified by investigating a transmission spectrum of water vapor.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.6
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• pp.1103-1108
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• 2023

In this paper, we propose a new algorithm for attenuating the background noises in acoustic signal. This algorithm improves the noise attenuation performance by using the FNN(: Full-connected Neural Network) deep learning algorithm instead of the existing adaptive filter after wavelet transform. After wavelet transforming the input signal for each short-time period, noise is removed from a single input audio signal containing noise by using a 1024-1024-512-neuron FNN deep learning model. This transforms the time-domain voice signal into the time-frequency domain so that the noise characteristics are well expressed, and effectively predicts voice in a noisy environment through supervised learning using the conversion parameter of the pure voice signal for the conversion parameter. In order to verify the performance of the noise reduction system proposed in this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed. As a result of the experiment, the proposed deep learning algorithm improved Mean Square Error (MSE) by 30% compared to the case of using the existing adaptive filter and by 20% compared to the case of using the STFT(: Short-Time Fourier Transform) transform effect was obtained.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.715-718
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• 2002

A control strategy for a dynamic system under Irregular disturbance by using stochastic controller is developed. In order to design stochastic controller. system dynamic model in real domain is transformed dynamic moment equation in stochastic domain by F-P-K approach. A study of real time control technique for stochastic controller is presented. The performance of stochastic controller is verified through experiment used by real time control technique method.

• Current Optics and Photonics
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• v.2 no.5
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• pp.413-421
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• 2018

We propose a novel methodology based on the multiplication function to improve the signal-to-noise ratio (SNR) for vibration detection in a phi optical time-domain reflectometer system (phi-OTDR). The extreme-mean complementary empirical mode decomposition (ECEMD) is designed to break down the original signal into a set of inherent mode functions (IMFs). The multiplication function in terms of selected IMFs is used to determine a vibration's position. By this method, the SNR of a phi-OTDR system is enhanced by several orders of magnitude. Simulations and experiments applying the method to real data prove the validity of the proposed approach.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4C
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• pp.418-424
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• 2007

This paper proposes a scheme to track channel response in OFDM systems with null sub-carriers. The proposed channel tracking scheme estimates the channel response first in the frequency domain by using the decision directed data. The time domain channel estimation is then performed to remove additive white Gaussian noise (AWGN) components further. Due to the channel estimation in the frequency domain, no inverse matrix calculation is required in the time domain channel estimation. Computational reduction in the proposed method is about 93%, compared with the conventional time domain channel estimation method. Mean square error (MSE) and bit error rate (BER) performances are evaluated by using computer simulation. The proposed method shows the same performance as that of the conventional time domain channel estimation even though the significant computational reduction.