• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.20 no.8
• /
• pp.1422-1430
• /
• 2016

Researches for oceans are limited to military purpose such as underwater sound detection and tracking system. Underwater acoustic communications with low-probability-of-interception (LPI) covert characteristics were received much attention recently. Covert communications are conducted at a low received signal-to-noise ratio to prevent interception or detection by an eavesdropper. This paper proposed optimal covert communication model based on direct sequence spread spectrum for underwater environments. Spread spectrum signals may be used for data transmission on underwater acoustic channels to achieve reliable transmission by suppressing the detrimental effect of interference and self-interference due to jamming and multipath propagation. The characteristics of the underwater acoustic channel present special problems in the design of covert communication systems. To improve performance and probability of interception, we applied BCJR(Bahl, Cocke, Jelinek, Raviv) decoding method and the direct sequence spread spectrum technology in low SNR. Also, we compared the performance between conventional model and proposed model based on turbo equalization by simulation and lake experiment.

• Journal of the Korean Society for Nondestructive Testing
• /
• v.15 no.1
• /
• pp.291-298
• /
• 1995

An ultrasonic testing uses the directivity of the ultrasonic wave which propagates in one direction. The directivity is expressed as the relationship between the propagate direction and its sound pressure. This paper studied the directivity of shear waves emitted from angle probes and scattered from surface defects by using visualization method. These experimental results were compared with the theory which was based on the continuous wave. The applicability of continuous wave theory was discussed in terms of the parameter $d/{\lambda}$; where d is transducer or defect size and ${\lambda}$ is the wavelength. In the case of angle probes, the experimental results show good agreement with theoretical directivity on the principal lobe. When defect size was smaller than the wavelengths, clear directivity in the reflected wave was observed. In the case of the same ratio of defect size to wavelength, the directivity of reflected waves from the defect show almost the same directivity in spite of frequency differences. When the $d/{\lambda}$ is greater than 1.5, measured directivities almost agreed with the theoretical one.

• Journal of Sensor Science and Technology
• /
• v.13 no.6
• /
• pp.479-487
• /
• 2004

The prevention of noise induced hearing loss is very important, because there is no treatment for it. There are some kinds of devices for hearing protection, and those are effective in preventing a noise induced hearing loss. However, people often resist the use of hearing protection devices because it is difficult to have a conversation. Therefore, a hearing protection device is must effective not only in hearing protection but also in preserving communication ability. In this paper, we proposed a small acoustic filter for ITE type noise protector to solving the problem about the difficulties of conversation. That is applied a principle of acoustic filters that have been used for a muffler of automobiles, guns, and etc. To find out the sound transmission characteristic at the eardrum, we regarded an acoustic filter and external ear canal as a coupled system. So, we simulated the coupled system with OrCad, and experimented with a designed acoustic filter and a 2 cc coupler which has the same transmission characteristic as the external ear canal has. We confirmed that it is possible to adjustment acoustic transmission characteristics through simulation of electrical model for acoustic filter and external ear and experiments using designed small acoustic filters.

• Journal of the Korea Society of Computer and Information
• /
• v.15 no.2
• /
• pp.109-117
• /
• 2010

Digital watermarking has drawn extensive attention for protecting digital contents from unauthorized copying. This paper proposes a new watermarking scheme in frequency domain for copyright protection of digital audio. In our proposed watermarking system, the original audio is segmented into non-overlapping frames. Watermarks are then embedded into the selected prominent peaks in the magnitude spectrum of each frame. Watermarks are extracted by performing the inverse operation of watermark embedding process. Simulation results indicate that the proposed scheme is robust against various kinds of attacks such as noise addition, cropping, resampling, re-quantization, MP3 compression, and low pass filtering. Our proposed watermarking system outperforms Cox's method in terms of imperceptibility, while keeping comparable robustness with the Cox's method. Our proposed system achieves SNR (signal-to-noise ratio) values ranging from 20 dB to 28 dB. This is in contrast to Cox's method which achieves SNR values ranging from only 14 dB to 23 dB.

• Perspectives in Nursing Science
• /
• v.14 no.1
• /
• pp.10-20
• /
• 2017

Purpose: The aim of this study was to explore the patients' perspectives on nursing students' clinical practices in the wards, and to investigate their willingness for allowing students to practice on them. Methods: This was a descriptive study. 116 inpatients were recruited from the S University Hospital. A 60-item questionnaire was applied to collect the data. The participants were 19 years and older with sound judgement, and were not in special or intensive care units. Data analysis was done in SPSS/WIN 22.0 using descriptive statistics, Fishers exact test, and the ANOVA test. the participant answered to questionnaire from April 29th 2016 to May 10th. Results: 40 participants (34.5%) stated they would allow students' practice, while 72 (61.2%) said they would allow only under staff supervision. 5 participants (4.3%) stated they would not allow whatsoever. The 3 most allowed were emotional support, oral care, and vital signs measurement while the 3 least allowed were gastric feeding, intravenous catheterization, and urinary catheterization. Conclusion: Patients were more inclined to allow students to practice on them when a member of the medical team was present. A fair number of participants said they would be more inclined to allow students' practice if they felt the student was competent; hence, reinforcing simulation sessions is vital in enhancing students' competency and ultimately practice allowance.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.1
• /
• pp.12-22
• /
• 2017

It is important to predict accurately reverberation level, which is a limiting factor in underwater target detection. Recently, the studies have been expanded from monostatic sonar to bistatic sonar in which source and receivers are separated. To simulate the bistatic reverberation level, the computation processes for propagation, scattering strength, and scattering cross section are different from those in monostatic case and more complex computation processes are required. Although there have been many researches for bistatic reverberation, few studies have assessed the bistatic scattering cross section which is a key factor in simulate reverberation level. In this paper, a new method to estimate the bistatic scattering cross section is suggested, which uses the area of intersection between two circles. Finally, the reverberation levels simulated with the scattering cross section estimated using the method suggested in this paper are compared with those estimated using the methods previously suggested and those measured from an acoustic measurements conducted in May 2013.

• Journal of Communications and Networks
• /
• v.18 no.3
• /
• pp.387-396
• /
• 2016

Multiuser communication has been an important research area of underwater acoustic communications and networking. This paper studies the use of adaptive orthogonal frequency-division multiple access (OFDMA) in a downlink scenario, where a central node sends data to multiple distributed nodes simultaneously. In practical implementations, the instantaneous channel state information (CSI) cannot be perfectly known by the central node in time-varying underwater acoustic (UWA) channels, due to the long propagation delays resulting from the low sound speed. In this paper, we explore the CSI feedback for resource allocation. An adaptive power-bit loading algorithm is presented, which assigns subcarriers to different users and allocates power and bits to each subcarrier, aiming to minimize the bit error rate (BER) under power and throughput constraints. Simulation results show considerable performance gains due to adaptive subcarrier allocation and further improvement through power and bit loading, as compared to the non-adaptive interleave subcarrier allocation scheme. In a lake experiment, channel feedback reduction is implemented through subcarrier clustering and uniform quantization. Although the performance gains are not as large as expected, experiment results confirm that adaptive subcarrier allocation schemes based on delayed channel feedback or long term statistics outperform the interleave subcarrier allocation scheme.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.40 no.1
• /
• pp.187-192
• /
• 2015

Underwater acoustic communication channels have very complex channel characteristics caused by time-varying sea surface, submarine topography, sound speed, and geometry between transmitter and receiver. Especially, the channel has time-variance and doppler effect due to wind and sea current. We have to recognize the channel state and apply it to communication technique for increasing transmission efficiency in the underwater acoustic channel. In this paper, we present the frame recursive modulation and demodulation method using ambiguity function and autocorrelation function to estimate the doppler frequency. Furthermore, we conducted the simulation and sea experiment to evaluate the performance of the proposed method. When the channel coding technique was not used, the bit error rate performance of the proposed method was improved about 32 % compared with conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.11C
• /
• pp.1573-1580
• /
• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Proceedings of the KOSOMBE Conference
• /
• v.1996 no.11
• /
• pp.240-242
• /
• 1996

Use of electrical stimulation for diagnosis and treatment of pyhsiolgocial impairment on the auditory system was reported by many researchers since 1800. Unfortunately, there were no standard methods or theoretical bakground for choosing stimulus conditions because of the lack of understanding on the transmission of sound through the auditory pathways. Therefore, if we could find the mechanisms of normal and abnormal functions for the system objectively, it may be possible to establish the conditions for the appropriate treatment This study was devised with three experimental conditions of electrical stimulation(0.1, 1, and 10mA) to verify the effect of electrical stimulation on the auditory system. ABRs and ECochGs were obtained from 3 Guinea Pigs before and after each stimulation, and five parameters were extracted from ABR and ECochG waveforms. The results showed that the mean differences of the first peaks of ABR, I, before and after all the stimulus conditions were significant at the significant level of p<0.05. Also, amplitude of ECochGs peak(AP) showed higher degrees of differences for the electrical stimulation than that of ABRs. Based on the results of this experiment establishment of an electrical model which characterizes the auditory system could be achieved by computer simulation. This will lead to the design of most accurate model for the real auditory system, which provide the quantitive information regarding diagnosis and treatment.