• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2006.05a
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• pp.114-117
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• 2006

The air filter in engine intake system has a function of filtrating the dirt in the scavenging air as well as attenuating the noise. The noise attenuation within the air cleaner filter, however, has been regarded as negligible by the field engineers. In this paper, for the analysis of the acoustical performance of air filter, an acoustical model was suggested and the characteristics of air filter system were investigated. Fibrous structure of the filter element was modeled as a micro-perforated panel using the flow resistivity and porosity. The pleated geometry of the filter element was modeled as two coupled ducts that have permeable walls, in which each duct area was assumed being constant. Using such simplified geometry, a mathematical model was developed for the sound propagation within a narrow duct system. Visco-thermal effect was considered in modeling the sound propagation through such tubes; the filter box was modeled as a rigid rectangular box. By combining two models, a four-pole transfer matrix was derived. For the validation purpose, transmission loss was measured for a plastic rectangular box containing an air filter. A noticeable effect of the air filter element was observed by including the filter into the box. Comparing the predicted and measured data, we found that the predicted TL agrees well with experimental results, in particular, in magnitude and frequency at TL troughs.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.670-672
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• 2014

In this study, we investigate active control of noise transmitted through a window of enclosures minimizing the acoustic power. To reduce noise of the enclosures, passive methods with absorbing material are generally used. The passive methods, however, are limited use due to the vantilation windows. In this case, these windows are path of noise leakage. Feedforward active noise control technology is applied to minimize the sound power from the enclosure. The feedforward controller is implemented with FIR filter based on the transfer functions calculated numerically. The controller reflects the delay due to FIR filter. The noise transmitted through the window is actively controlled, and the reduction of the power is obtained by 15dB.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2151-2156
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• 2006

Beamforming for underwater acoustic communication (UAC) is affected by the broadband feature of UAC signal, which has relatively low currier frequency as compared to the signal bandwidth. The narrow-band assumption does not hold good in UAC. In this paper, we discuss a broadband FIR beamformer for UAC using the baseband equivalent way signal model. We consider the broadband FIR beamformer for QPSK UAC with carrier frequency 25kHz and symbol rate 5kHz. Array geometry is a uniform linear way with 8 omni-directional elements and sensor spacing is the half of the carrier wavelength. The simulation results show that the broadband n beamformer achieves nearly optimum signal to interference and noise ratio (SINR) and outperforms the conventional narrowband beamformer by SINR 0.5dB when two-tap FIR filter is employed at each sensor and the inter-tap delay is a quarter of the symbol interval. The broadband FIR beamformer performance is more degraded as the FIR filter length is increased above a certain value. If the inter-tap delay is not greater than half of the symbol period, SINR performance does not depend on the inter-tap delay. More training period is required when the inter-tap delay is same as the symbol period.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.987-994
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• 2020

In this paper, we studied the effect of kernel size of CNN layer on performance in acoustic noise attenuators. This system uses a deep learning algorithm using a neural network adaptive prediction filter instead of using the existing adaptive filter. Speech is estimated from a single input speech signal containing noise using a 100-neuron, 16-filter CNN filter and an error back propagation algorithm. This is to use the quasi-periodic property in the voiced sound section of the voice signal. In this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed to verify the performance of the noise attenuator for the kernel size. As a result of the simulation, when the kernel size is about 16, the MSE and MAE values are the smallest, and when the size is smaller or larger than 16, the MSE and MAE values increase. It can be seen that in the case of an speech signal, the features can be best captured when the kernel size is about 16.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.309-316
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• 2012

This paper suggest configuration method of small-size system and design method of module using multi power. In particular module designed to focus on PDN(Power Distribution Network) and filter configuration method to improve the characteristics of noise sensitive analog IC. For the prevention of high-voltage noise, manufactured module is used the ground-isolated technique and via stitching, and is connected grounds with a series of filters. In this paper, so we analyzed PDN structure through the simulation using lumped modeling and confirmed through measurement. Simulation results, when connecting 4.7uH inductor, we made certain that noise of -7dB decreases much more than when it did not. And it was confirmed 12% less than Background Noise.

• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.570-575
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• 2004

In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.732-739
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• 2009

Flow phantom with stenosis was manufactured using an auto-injector to obtain angiostenotic flow information and quality assurance (QA) for ultrasound diagnostic instrumentation. Effectiveness of manufactured flow phantom with stenosis was investigated with power Doppler that was known to have diagnostic efficiency for angiostenosis. The flow phantom with stenosis was manufactured to 70% stenosis with 8 mm and 2.4 mm silicon tube, and silicone tube was covered with gelatin that has acoustic characteristics similar to soft tissue. When the linear transducer was used for measurement, the estimated diameter of normal vessel was measured lower than that of normal value, and the estimated diameter of stenosed vessel was measured higher than that of normal value. The measured parameters were not affected except for the radical conditions such as gain of 60%, PRF of 3000 Hz, use of maximal filter or angle. In addition, when the convex transducer was used for measurement, measurement parameters were affected by gain, PRF, filter, and angle. Therefore it is expected that flow phantom with stenosis manufactured with an auto-injector will be utilized effectively for QA of angiostenotic diagnosis.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.5-13
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• 1994

Simulation of the target-scattered echo requires the understanding of scattering mechanism at the highlight points. In this paper, the basic assumption of Highlight Model is reviewed through the analyzed data obtained in the acoustic water tank experiment. The analysis shows that the scattering mechanism involves pulse elongation and frequency shift as elements of target-scattered echo, and that the internal structures affect the temporal response of the target-scattered echo significantly. The band-limited impulse response or Green's function due to the diffraction from highlight points of internal structures is not mere delta function, but acts like a filter, which causes frequency shift and is elongated in time.