• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.2
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• pp.230-235
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a performance of perceptual filter using intelligent estimation of noise pattern from a band degraded by additive noise. The proposed method doesn't use the estimated noise which is obtained from silent range. Instead new estimated noise according to the power of signal and effect of noise variation is considered for each frame. So the noisy audio signal is enhanced by the method which controls a estimation of noise Pattern effectively in a noise corruption band. To show the performance of the proposed algorithm, various input signals which had a different signal-to-noise ratio(SNR) such as $5\cal{dB},\;10\cal{dB},\;15\cal{dB}\;and\;20\cal{dB}$ were used to test the proposed algorithm. we carry out SSNR and NMR of objective measurement and MOS test of subjective measurement. An approximate improvement of $7.4\cal{dB},\;6.8\cal{dB},\;5.7\cal{dB},\;5.1\cal{dB}$ in SSNR and $15.7\cal{dB},\;15.5\cal{dB},\;15.2\cal{dB},\;14.8\cal{dB}$ in NMR is achieved with the input signals, respectively. And we confirm the enhancement of tone quality in terms of mean opinion score(MOS) test which is result of subjective measurement.

• Journal of the Korea Society of Computer and Information
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• v.25 no.9
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• pp.91-100
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• 2020

This paper presents an evaluation framework for prototyping multimedia streaming services including audio and video in a distributed and/or decentralized storage that can evaluate service quality and performance under various network conditions. The evaluation framework focuses on important indicators which measure and improve service quality by applying decentralized storage to multimedia streaming services that can mimic the scalability of the existing server-client software architecture and the issue of a single point of failure. The integrated framework not only measures performance indicators for evaluating the quality and performance of multimedia streaming on open source based multimedia content streaming services, but also adjusts network quality using network virtualization technology for comprehensive evaluations. The experimental results show that the integrated framework has low overhead in building and operating a decentralized storage with multimedia streaming services on a single host computer which validates the scalability of the developed framework.

• Journal of The Korean Association For Science Education
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• v.29 no.4
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• pp.400-413
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• 2009

The purpose of this study was to investigate middle school students' processes of argumentation in science lessons and to compare students' argumentation in different contexts (socioscientific context/scientific context). An argumentation-based teaching-learning strategy was used to enhance quality in students' arguments in science lessons. Data were collected from five lessons by video- and audio-recording eight groups of four students each engaging in argumentation. The quality and frequency of students' argumentation was analyzed using an assessment framework based on the work of Toulmin. The findings showed that: (a) there was improvement in the quality of students' argumentation; (b) there were no differences in the structure of argumentation and percentage of explanatory argumentation components as well as dialogic argumentation components in different argumentation contexts. The results of this study showed that students' argumentation can be enhanced with strategic argumentation teaching-learning.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.1012-1021
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• 2005

This paper describes the design and implementation of an efficient multi-point multimedia conference system using IP grouping. Existing multi-point multimedia conference systems are difficult for multi-user to perform efficient cooperation due to bandwidth limitation for data transmission of video, audio and documentation. In the case that multi-user uses limited bandwidth, smooth cooperation does not accomplish due to transmission delay for the real-time transmission of image and speech data. A hybrid transfer method which is mixed with distributed and centralized methods is used for smooth cooperation, and the network bandwidth is reduced by forming multi-user conference systems of IP grouping in this paper. Also, adaptive image frame variations are used to solve bottleneck effect according to the number of users. An efficient multi-user conference system is designed to support audio quality.

• Journal of Advanced Navigation Technology
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• v.15 no.5
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• pp.714-721
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• 2011

In this paper, developing management programs for EMI tracking can navigate the site quickly and solve EMI tracking cause and location to use materials such as analysis of air accidents, EMI site location data of 1000 RF companys, radio wave spectrum analysis and audio data. these data are databased and used comparable data. Also, EMI has been prevented by establishing continuous monitoring system through a 24-hour surveillance. Therefore we were able to provide high quality air waves in order to prevent aircraft accidents. In addition, radar control staff of Korea Airports Corporation against passenger aircraft that will prevent the worst aircraft accident have been established based to continue periodic aviation frequency protection and Portable Electronic Devices(PED) on board aircraft to prevent the culture of safety campaign.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.8
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• pp.1069-1082
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• 1993

The International Organization for Standardization(ISO) has undertaken an effort to develop a standard for video and associated audio on digital storage media. This effort is known by the name of the expert group that started if : MPEG-Moving Picture Experts Group Is currently part of the ISO-I EC/J TC1/SC2/WG11. The promise of MPEG-2 is that a video signal and its associated audio can be compressed to a bit rate of about 10 Mbits/s with an acceptable quality. In this paper, the implementation of a video compression simulator based on MPEG-2 Video Test Model 2(TM2) is described and analyzed according to the simulation results. The implemented simulator is also applied to code HDTV sequences at the several bit rates. Some computer simulation results using the MPEG and the HDTV test sequences are given. In addition, some techniques which can improve the coding efficiency of the implemented video compression simulator are also suggested.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.

• Convergence Security Journal
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• v.8 no.4
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• pp.41-49
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• 2008

VoIP is a service that changes the analogue audio signal into a digital signal and then transfers the audio information to the users after configuring it as a packet; and it has an advantage of lower price than the existing voice call service and better extensibility. However, VoIP service has a system structure that, compared to the existing PSTN (Public Switched Telephone Network), has poor call quality and is vulnerable in the security aspect. To make up these problems, TLS service was introduced to enhance the security. In practical system, however, since QoS problem occurs, it is necessary to develop the VoIP security system that can satisfy QoS at the same time in the security aspect. In this paper, a user authentication VoIP system that can provide a service according to the security and the user through providing a differential service according to the approach of the users by adding AA server at the step of configuring the existing VoIP session is suggested. It was found that the proposed system of this study provides a quicker QoS than the TLS-added system at a similar level of security. Also, it is able to provide a variety of additional services by the different users.

• The Journal of the Acoustical Society of Korea
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• v.39 no.4
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• pp.292-302
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• 2020

Machine learning algorithms have made immense contributions in various fields including sonar and radar applications. Recently developed Cycle-Consistency Generative Adversarial Network (CycleGAN), a variant of GAN has been successfully used for unpaired image-to-image translation. We present a modified CycleGAN for translation of underwater ship engine sounds with high perceptual quality. The proposed network is composed of an improved generator model trained to translate underwater audio from one vessel type to other, an improved discriminator to identify the data as real or fake and a modified cycle-consistency loss function. The quantitative and qualitative analysis of the proposed CycleGAN are performed on publicly available underwater dataset ShipsEar by evaluating and comparing Mel-cepstral distortion, pitch contour matching, nearest neighbor comparison and mean opinion score with existing algorithms. The analysis results of the proposed network demonstrate the effectiveness of the proposed network.