• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.

• The Journal of the Acoustical Society of Korea
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• v.9 no.2
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• pp.53-60
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• 1990

MSK waveforms are advantageous from the point of view of reduced interchannel interference. MSK waveform, however, is more difficult to generate than PSK due to its complex pulse shape. A SAW filter is ideally suited for generation of MSK waveform because it provides accurate control of the required impulse responese and linear phase at high frequncies. In this paper, an implementation of SAW based PSK-MSK convert filter, which can easily convert the high frequency PSK waveform to MSK, is studied.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.42 no.4 s.304
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• pp.45-50
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• 2005

In digital hearing aid system, one of the main problems is acoustic feedback which is known as howling because of miniaturization md high-gain amplification. In this paper, we proposed a feedback cancellation algorithm for hearing aid using time delay and time-varying decorrelation filter. The proposed algorithm has a kind of adaptive filter structure, which is combined with time delay and time-varying decorrelation filter to improve feedback cancellation. An all pass filter was implemented as the time-varying decorrelation filter using low frequency modulator. From the result of computer simulation, it is verified that the proposed algorithm has good ability to cancel feedback.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

• Journal of Navigation and Port Research
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• v.28 no.6
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• pp.491-496
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• 2004

The paper describes measurement techniques for an acoustic transfer characteristic of ship's bridge stimulated by a whistle sound The response sounds, according to the opening-shutting conditions of bridge doors for Training Ship ‘SAENURI’, are measured by B＆K 2260D equipment, and then the frequency responses are extracted by B＆K 7830 software. To evaluate the measured transfer characteristic, the 128th order FIR (Finite Impulse Response) filters, containing the different frequency characteristics, are constructed based on the frequency sampling-based design method Using evaluation indexes with six scales, psychological assessments by five subjects are carried out with test sounds which are obtained from convolving the source signal with FIR filters. As results of tests, the test sounds gives $A_S$ 3.3∼4.7 which means the psychological sense of ‘it is almost similar sound as original ones in a real world’, and thus it is clearly seen that the proposed method can be used for the measurement of an acoustic transfer characteristic of ship’s bridge.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.55-60
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• 2004

The paper describes measurement techniques for an acoustic transfer characteristic of ship's bridge stimulated by a whistle sound. The response sounds, according to the opening-shutting conditions of bridge doors for T／S ‘SAENURI’, are measured by B＆K 2260D equipment, and then extracted the frequency responses by B＆K 7830 software. To evaluate the measured transfer characteristic, the 128th order FIR (Finite Impulse Response) filters, containing the frequency characteristic, are constructed based on the frequency sampling-based design method. Using evaluation indexes with six scales, psychological assessments by five subjects are carried out with test sounds which are convolved source sound and FIR filters. As results of tests, the test sounds gives A／sub s／=3.3∼4.7 which means the psychological sense of 'it is almost similar sound as original ones in a real world', and thus it is cleary seen tint the proposed method can be suit for the measurement of an acoustic transfer characteristic of ship's bridg.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.3
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• pp.570-575
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• 2019

Active muffler is implemented by applying active noise control technique to reduce exhaust noise of automobile muffler. Conventional Filtered_x LMS algorithm has a problem that the degree of control filter becomes very large and convergence deteriorates when acoustic feedback is present. The recursive LMS algorithm can compensate for this problem because it can be easily diverted in the adaptive filter adaptation process. In this paper, the structure of the primary path and the secondary path transfer function is designed as the IIR filter to improve the convergence performance and the computational burden, and the stabilization filter algorithm is applied to secure stability which is a disadvantage of the IIR filter structure. The stabilization filter algorithm plays a role of pulling the pole into the unit circle to prevent the pole of the transfer function corresponding to the acoustic feedback from diverging during the adaptation process. In this way, the computational burden of the active muffler system and the convergence performance can be improved. In order to show the usefulness of the proposed system, we compared the performance of the proposed Filtered_x LMS algorithm with the performance of the proposed system for the exhaust sound of a diesel engine, which is a variable environment. Compared to conventional algorithm, proposed algorithm's computational burden is less than half, and convergence performances are more than 4 times.

• Journal of the Korea Convergence Society
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• v.11 no.11
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• pp.249-255
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• 2020

This study developed the health monitoring method of press process using the acoustic emission (AE) sensor and high-pass filter. Also, the AE parameters such as ring-down count and peak amplitude are used. Based on this AE signal, the AE parameters were acquired and was utilized to detect the crack of the specimen. Since the defect detection is difficult due to noise and low magnitude of signal, the signal noise and press operation frequency were checked through the Short Time Fourier Transform(STFT) and damped. High-pass Filtering data was applied to AE parameters to select effective parameters. By using this signal processing techniques, the proposed AE parameters could improve the performance of defect detection in the press process.

• The Journal of the Acoustical Society of Korea
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• v.39 no.1
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• pp.1-7
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• 2020

The acoustic parabolic equation method in the ocean is an efficient technique to calculate the acoustic field in the range-dependent environment, emanating from a point source. However, we often need to use the directional source with a main beam in the practical problem. In this paper, we present two methods to implement the directional source in the acoustic parabolic equation code easily. One is simply to filter the Delta function idealized as an omni-directional point source. Another method is based on the rational filtering of the self-starter solution. It has a limitation not to separate the up-going and the down-going wave for the depth, but would be useful in implementing the mode propagation. Numerical examples for validation are given in the Pekeris environment and the deep sea environment.

• Journal of Korea Multimedia Society
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• v.3 no.6
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• pp.604-610
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• 2000

A stereo acoustic echo canceller has significant cross-correlation between each channel signal. As the result, the adaptive filter coefficient that estimates the acoustic echo path of a receiving room can misconverge to the path or converge slowly. In this paper, a new preprocessor using absolute difference in stereo signals is proposed to reduce cross-correlations and to improve the removal efficiency of the stereo acoustic echo. Compared to the previous preprocessor using a half wave rectifier, the newly proposed preprocessor showed better performance according to computer simulation. In addition, when the paths of transmitting room were changed the performance of the proposed preprocessor was not degraded.