• Journal of the Korean Society for Nondestructive Testing
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• v.31 no.2
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• pp.165-173
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• 2011

The acoustic emission(AE) technique is a well established method to carry out structural health monitoring(SHM) of large structures. However, the real-time monitoring of the crack growth in the roller coaster support structures is not easy since the vehicle operation produces very large noise as well as crack growth. In this investigation, we present the waveform parameter filtering method to classify acoustic sources in real-time. This method filtrates only the AE hits by the target acoustic source as passing hits in a specific parameter band. According to various acoustic sources, the waveform parameters were measured and analyzed to verify the present filtering method. Also, the AE system employing the waveform parameter filter was manufactured and applied to the roller coaster support structure in an actual amusement park.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.84-90
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• 2000

The bearing detection of radiated target noise is very important at underwater acoustic measurement and passive detection. It differs the arrival tines of received signal at each sensor. Therefore, the bearing can be obtained from the time delay. This paper proposes a new algorithm using the RLSL adaptive filter for TDE. The proposed method is particularly attractive when there is a limitation of priori information about the received signal spectra and when the delay is subject to variation. As the simulation results, it is shown that the proposed algorithm has better convergence characteristics and TDE speed, and so that the usefulness of proposed algorithm is confirmed.

• Korean Journal of Optics and Photonics
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• v.11 no.6
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• pp.423-428
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• 2000

We have demonstrated a -8 dB additional reduction in the intensity sidelobe of an apodized-interaction-strength guide-wave acousto-optic filter with a center passband of 1551.6 nm. Acoustic-intensity weighting was achieved by launching a surface acoustic wave (SAW) beam in a straight acoustic waveguide, and gradually transferring this SAW intensity to the active device, and back out, by evanescent-wave coupling across a 50 !lm barrier over a 19 rom interaction length. The intensity sidelobe was -4.27 dB for an unapodized filter with abmpt onset and cutoff of the interaction, but sidelobes were reduced to at most -12.68 dB for a SAW intensity with raised-cosine weighting. The RF driving power was 17.78 mW. A linear tuning rate of 8.86 nmIMHz and a spectral width of -1.7 nm were demonstrated. rated.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.491-498
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• 2009

This paper presents the design method of an impedance matching network using an isolation transformer and the Chebyshev filter function for the high efficiency and the flat power driving of an underwater acoustic piezoelectric transducer. The proposed impedance matching network is designed for minimizing the reactance component of transducer and having the flat power response in the wide frequency range. We design a low pass filter with ladder-type circuit using the Chebyshev function as standard prototype filter function. In addition, we design the impedance matching network which is suitable for the equivalent circuit of transducer and the turn ratio of transformer through the bandpass frequency transformation. The proposed method is applied to the simulated dummy load of the tonpilz-type transducer operating in the middle frequency range. The simulation results are compared with the measured characteristics and the validity of the proposed method is verified.

• Journal of Korea Multimedia Society
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• v.10 no.2
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• pp.200-208
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• 2007

The noise induced hearing loss, which is one type of the hearing losses, is well known occupational diseases. The prevention of the noise induced hearing loss is very important, because it is unrecoverable. There are some kinds of devices for hearing protection, and those are effective in preventing the noise induced hearing loss. However, people often resist the use of hearing protection devices because they have some difficulties during conversation in wearing the devices. To solve the problem, a small acoustic filter can be used for a hearing protection device. In this paper, we designed several kinds of small acoustic filters and analyzed the characteristics of designed small acoustic filters using Finite Element Analysis (FEA). From FEA results, we proved the characteristics of designed small acoustic filters which vary according to length and diameter. And we found out that the general tendencies of modeled filters using FEA are same as acoustic experiment results of designed filters.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.303-313
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• 2010

This paper describes an acoustic communication receiver structure, which is designed for QPSK (Quadrature Phase Shift Keying) signal with 25 kHz carrier frequency and 5 kHz symbol rate, and takes samples from received signal at 100 kHz sampling rate. Based on the described receiver structure, optimum design parameters, such as number of taps of FF (Feed-Forward) and FB (Feed-Back) filters and forgetting factor of RLS (Recursive Least-Square) algorithm, of joint equalizer are determined to minimize the BER (Bit Error Rate) performance of the joint equalizer output symbols when the acquisition data in shallow water using implemented acoustic transducers is decimated at a rate of 2:1 and then enforced to the input of receiver. The transmission distances are 1.4 km, 2.9 km, and 4.7 km. Analysis results show that the optimum number of taps of FF and FB filters are different according to the distance between source and destination, but the optimum or near optimum value of forgetting factor is 0.997. Therefore, we can reach a conclusion that the proper receiver structure could change the number of taps of FF and FB filters with the fixed forgetting factor 0.997 according to the transmission distance. Another analysis result is that there are an acceptable performance degradation when the 16-tap-length simple filter is used as a low-pass filter of receiver instead of 161-tap-length matched filter.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.24-29
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• 2004

In ocean acoustic tomography (OAT), the pulse compression techniques using M-sequences are employed in the many studies for investigating the ocean structures. M-sequences can provide the good time and Doppler resolution in the process of demodulation using matched-filter. The signal-to-noise (SNR) performance at the output of receiver may be improved by manipulating received signal, i. e. coherently averaging. The processing time can be significantly reduced by using fast hadarmard transform (FHT) or fast Fourier transform (FFT). In this paper, we estimate the multipath arrival structures and delay times using the East Korean Sea experiment data and explore the compensation method for the detrimental effects on performance due to sampling rate error. We also analyze the characteristics of the ocean acoustic channels through scattering function, delay power profile, and time dispersions.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.57-63
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• 1997

This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity.

• Proceedings of the Korean Society of Propulsion Engineers Conference
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• 2003.10a
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• pp.153-156
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• 2003

Backhole, which is one of geometric parameters in swirl coaxial injectors, is found to affect the inner flow motion and the acoustic characteristics of the swirl injector. In order to analyze the effect of the backhole as a damping device such as acoustic cavities of the combustion chamber, it was regarded as a Helmholtz or Quarter-wave resonator. As a result, it is known that the swirl coaxial injector with the backhole may produce the resonant frequency coincided with the frequency of the combustion chamber.