• The Sea:JOURNAL OF THE KOREAN SOCIETY OF OCEANOGRAPHY
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• v.4 no.3
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• pp.170-179
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• 1999

The SUS (Signal, Underwater Sound)-OAT experiment was carried out in the Ulleung Basin of the East Sea on 3 June 1997. The SUS-OAT system consisted of aircraft deployed shots as sources and a vertical line array (VLA) tethered by a receiver ship was used to survey a large area where a mesoscale warm eddy appears frequently. The experiment was carried out such that explosive charges set to detonate at 800 ft depth were dropped in a rectangular ($120{\times}120$ km). Sources were a rapidly deployable SUS charge (MK 61 MOD 0), and receiver is a fixed VLA, 90 m in length (150-240 m in receiver depth), composed of 10 elements equally spaced. The reference ray paths are computed by range-dependent acoustic model in canonical ocean based on the historical data. The singular value decomposition (SVD) method is used to obtain the horizontal perturbation of the temperature fields. Horizontal distributions of temperature fields at 150 m and 200 m depth show a weak warm eddy observed by AXBT and the inversely estimated temperature shows similar patterns in terms of the location of the warm eddy. In conclusion, the SUS-OAT experiment has been successful to estimate the position of warm eddy and its temperature field in the East Sea of Korea.

• Journal of Sensor Science and Technology
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• v.19 no.5
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• pp.361-368
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• 2010

ACROSS device, which is composed of an implantable microphone, a signal processor, and a vibrating transducer, is a fullyimplantable middle ear hearing device(F-IMEHD) for the recovery of patients with hearing loss. And since a microphone is implanted under skin and tissue at the temporal bones, the amplitude of the sound wave is attenuated by absorption and scattering. And the vibrating transducer attached to the ossicular chain caused also the different displacement from characteristic of the stapes. For the gain control of auditory signals, most of implantable hearing devices with the digital audio signal processor still apply to fitting rules of conventional hearing aid without regard to the effect of the implanted microphone and the vibrating transducer. So it should be taken into account the effect of the implantable microphone and the vibrating transducer to use the conventional audio fitting rule. The aim of this study was to measure gain characteristics caused by the implanted microphone and the vibrating transducer attached to the ossicle chains for the gain compensation of ACROSS device. Differential floating mass transducers (DFMT) of ACROSS device were clipped on four cadaver temporal bones. And after placing the DFMT on them, displacements of the ossicle chain with the DFMT operated by 1 $mA_{peak}$ current was measured using laser Doppler vibrometer. And the sensitivity of microphones under the sampled pig skin and the skin of 3 rat back were measured by stimulus of pure tones in frequency from 0.1 to 8.9 kHz. And we confirmed that the microphone implanted under skin showed poorer frequency response in the acoustic high-frequency band than it in the low- to mid- frequency band, and the resonant frequency of the stapes vibration was changed by attaching the DFMT on the incus, the displacement of the DFMT driven with 1 $mA_{rms}$ was higher by the amount of about 20 dB than that of cadaver's stapes driven by the sound presssure of 94 dB SPL in resonance frequency range.

• Science of Emotion and Sensibility
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• v.23 no.4
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• pp.105-116
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• 2020

The purpose of this study is to progress the limitations and disadvantages of existing safety clothing by applying high technology to current safety clothing that is produced and distributed only with fluorescent fabrics and retroreflective materials. Therefore, the industrial suspender-type safety belt and engineering technology are introduced, designed, and fabricated to help save a life in an emergency. First, the suspender-type safety belt to be developed is designed to emit light by LED attached to the film, and the body of the belt-wearer is recognized from a distance through retroreflection from the flashing LED. It aims to support people's safety by preventing accidents during roadside work, rescue activities, and sports activities at night. Second, with the development of advanced devices when the user is in an unconscious state due to distress or falls into an unconscious state due to distress or accident, the tilt sensor of the control unit attached to the belt automatically detects the angle of the human body and generates light and sound. It is intended to further enhance the utilization by mounting a sensing and signaling device that generates a distress signal and shaping it in the form of a belt attached to a vest that can be easily detached from the outside of the garment. When the wearer falls due to an accident, the tilt sensor of this belt detects the angle change and then the controller generates a high-frequency sound and repeated LED blinking signals at the same time. In the case of conventional safety vests, it is almost impossible to detect that the person is wearing a vest when there is no ambient light, but in case of the safety belts in this study, the sound and light signals of the safety belt enable us to find the wearer within 100 meters even when there is no ambient light.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.66-76
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• 2000

Acoustic performance of loudspeakers for sound reproduction has been qualitatively evaluated by using the listening test by juries in the development and final evaluation stages. However, the subjective evaluation method has many problems in the viewpoint of reliability and repeatability that are mainly related to the jury performance, as well as time and economy. In this reason, objective techniques should be tried to evaluate the acoustic performance of loudspeakers as well as the conventional subjective test. The object of this study is to find if there is any correlation between the statistically treated in test results and the measured results based on the loudness of reproduced sound signals. For the four-step statistical analysis, the analysis of variance (ANOVA) and Tukeys method are employed for dealing with the data from the listening test. For the objective evaluation, Zwickers loudness considering the human hearing characteristics is calculated for the measured sound signal emitted from each loudspeaker and the objective ratings such as fidelity rating (FR) and softness rating (SR) is suggested. The correlation between two ratings has been demonstrated for an actual set of loudspeakers using FR, SR and correlation coefficient. The method in this study can be useful in statistically evaluating commercial or prototype loudspeakers without using very time-consuming and expensive subjective testing.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.71-79
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• 2016

This study is intended to compare the performances of sound source localization methods used for stable automobile control by improving voice recognition rate in automobile environment and suggest how to improve their performances. Generally, sound source location estimation methods employ the TDOA algorithm, and there are two ways for it; one is to use a cross correlation function in the time domain, and the other is GCC-PHAT calculated in the frequency domain. Among these ways, GCC-PHAT is known to have stronger characteristics against echo and noise than the cross correlation function. This study compared the performances of the two methods above in automobile environment full of echo and vibration noise and suggested the use of a median filter additionally. We found that median filter helps both estimation methods have good performances and variance values to be decreased. According to the experimental results, there is almost no difference in the two methods' performances in the experiment using voice; however, using the signal of a song, GCC-PHAT is 10% more excellent than the cross correlation function in terms of the recognition rate. Also, when the median filter was added, the cross correlation function's recognition rate could be improved up to 11%. And in regarding to variance values, both methods showed stable performances.

• The Journal of the Acoustical Society of Korea
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• v.40 no.2
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• pp.139-147
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• 2021

This paper introduces and implements a Sound Event Detection (SED) system based on weakly-supervised learning where only part of the data is labeled, and analyzes the effect of parameters. The SED system estimates the classes and onset/offset times of events in the acoustic signal. In order to train the model, all information on the event class and onset/offset times must be provided. Unfortunately, the onset/offset times are hard to be labeled exactly. Therefore, in the weakly-supervised task, the SED model is trained by "strongly labeled data" including the event class and activations, "weakly labeled data" including the event class, and "unlabeled data" without any label. Recently, the SED systems using the mean-teacher model are widely used for the task with several parameters. These parameters should be chosen carefully because they may affect the performance. In this paper, performance analysis was performed on parameters, such as the feature, moving average parameter, weight of the consistency cost function, ramp-up length, and maximum learning rate, using the data of DCASE 2020 Task 4. Effects and the optimal values of the parameters were discussed.

• Journal of the Korean earth science society
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• v.42 no.6
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• pp.688-699
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• 2021

On September 23, 2020, at 1:39 a.m., a bright fireball above Seocheon was observed across the country. Two fireball explosions were identified in the images of the All-Sky Camera (ASC), and the shock waves were recorded at seismic and infrasound stations in the southwestern Korean Peninsula. The location of the explosion was estimated by a Bayesian-based location method using the arrival times of the fireball-associated seismic and infrasound signals at 17 stations. Realistic azimuth- and rang-dependent propagation speeds of sound waves were incorporated into the location method to increase the reliability of the results. The location of the sound source was found to be 36.050°N, 126.855°E at an altitude of 35 km, which was close to the location of the second fireball explosion. The two explosions were identified as sequential infrasound arrivals at local infrasound stations. Simulations of waveforms for long ranges explain the detection results at distant infrasound stations, up to ~266 km from the sound source. The dominant period of the signals recorded at five infrasound stations is about 0.4 s. A period-energy relation suggests the explosion energy was equivalent to ~0.3 ton of TNT.

• The Journal of the Acoustical Society of Korea
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• v.41 no.4
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• pp.435-445
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• 2022

In this study, the reverberation of a bistatic sonar operated in southeastern coast in the East Sea in July 2020 was analyzed. The reverberation sensor data were collected through an LFM sound source towed by a research vessel and a horizontal line array receiver 1 km to 5 km away from it. The reverberation sensor data was analyzed by various methods including geo-plot after signal processing. Through this, it was confirmed that the angle reflected from the sound source through the scatterer to the receiver has a dominant influence on the distribution of the reverberation sound, and the probability distribution characteristics of bistatic sonar reverberation varies for each beam. In addition, parametric factors of K distribution and Rayleigh distribution were estimated from the sample through moment method estimation. Using the Kolmogorov-Smirnov test at the confidence level of 0.05, the distribution probability of the data was analyzed. As a result, it could be observed that the reverberation follows a Rayleigh probability distribution, and it could be estimated that this was the effect of a low reverberation to noise ratio.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.21 no.5
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• pp.69-76
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• 2020

The sound signal radiated from an underwater source is received by the hydrophone of the system, including multi-path time-delay and multi-path signal by sea surface and bottom reflection. The system using a time-delay between received signals for the source localization shows performance degradation due to incoherence by the multi-path propagation environment and the disturbance of a marine environment. Various types of array and signal processing have been used for robust source range and bearing estimation in this environment. In this paper, we use a line array composed of doublet array and an estimated time-delay correction method for robust localization performance in a multi-path propagation environment. Three doublet arrays are located on the same line, and the time-delay between signals received on each doublet array is estimated in a two-step procedure. The estimated time-delay value is obtained by the cross-correlation function and corrected by the interaction formula between the center-frequency of received signal and the geometry of the array with respect to aperture. By this proposed procedure, the range and bearing of source from array were calculated. In order to confirm the validity of the proposed method and array, we simulated localization and estimation using the Monte-Carlo method.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.5
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• pp.239-246
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• 2023

In this paper, we propose an open-loop algorithm to classify speech and music signals using the spectral flux parameters and Mel Frequency Cepstral Coefficients(MFCC) parameters for the audio coder. To increase responsiveness, the MFCC was used as a short-term feature parameter and spectral fluxes were used as a long-term feature parameters to improve accuracy. The overall voice/music signal classification decision is made by combining the short-term classification method and the long-term classification method. The Gaussian Mixed Model (GMM) was used for pattern recognition and the optimal GMM parameters were extracted using the Expectation Maximization (EM) algorithm. The proposed long-term and short-term combined speech/music signal classification method showed an average classification error rate of 1.5% on various audio sound sources, and improved the classification error rate by 0.9% compared to the short-term single classification method and 0.6% compared to the long-term single classification method. The proposed speech/music signal classification method was able to improve the classification error rate performance by 9.1% in percussion music signals with attacks and 5.8% in voice signals compared to the Unified Speech Audio Coding (USAC) audio classification method.