• Proceedings of the Korea Information Processing Society Conference
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• 2003.11b
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• pp.1121-1124
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• 2003

스케줄링은 네트워크상의 공유된 자원을 보다 효과적으로 이용하기 위한 것으로서 delay, delay jitter, packet loss rate, throughput과 같은 서비스 값들의 QoA(Quality of Service)를 보장하기 위한 핵심 요소이다. 유선망에서의 스케줄링은 이미 익숙한 영역으로 많은 발전이 되어왔지만, 무선채널의 불안정성이나 사용자의 움직임으로 인해 발생하는 다양한 링크 에러율과 용량 때문에 무선망에서의 직접적용은 많은 문제를 일으키게 된다. 이 논문에서는 기존에 나와 있는 여러 무선 스케줄링 기법중 채널 보상 모델을 이용하여 서비스 차별화(Service Differentiation)와 공평성(Fairness)에 초점을 맞춘 QoS제공 성능향상 스케줄링 알고리즘을 제안하였다.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.30A no.11
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• pp.1-10
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• 1993

A Study on New Digital In this paper, we propose a new Angular Form Carrier Recovery Loop(AFCR loop) for PSK modulation technique. AF CR loop includes detected angle symbol and Multi Level Hardlimiter. Using zero crossing DPLL, we model 1st 2nd AF CR loop, and also derive SCurve. In order to prove steady state operation of AF CR loop, we evaluate performance of this loop by Monte-Carlo and analytical simulation method. We also compare the performance of AF CR loop to that of other loop in terms of acquisition, S-Curve, and RMS jitter. From the comparison result, we verify that the performance of AF CR loop operates well in steady state.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.11 no.1
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• pp.98-103
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• 2000

Background and Objectives : The etiology and pathophysiology of spasmodic dysphonia is yet unknown. This study was performed to determine if any laryngeal aerodynamic parameter distinguish the voice of patient diagnosed as having adductor spasmodic dysphonia from individuals with normal voice production and to investigate the pathophysiology of spasmodic dysphonia. Materials and Methods : fifteen women diagnosed as having adductor spasmodic dysphonia and fifteen normal control women participitated in this study Maximum phonation time, mean air flow rate, subglottic pressure, vocal efficiency, Vfo, NHR, VTI, FTRI, ATRI, Jitter percent, Shimmer percent were obtained from the participants using 'MDVP(multi-dimensional voice program)' of CSL(Computerized Speech lab, Kay Elemetrics, Co., Model No. 4300), and 'maximum sustained phonation' and 'IPIPI test' of AP II(Aerophone II, Kay Elemetrics, Co., Model 6800). Results : T-test statistical analysis revealed statistically different values for vocal efficiency, Vfo, NHR, MPT, litter percent, Shimmer percent between the spasmodic dysphonia group and the control group. Conclusions : Spasmodic dysphonia affects the ability of the laryngeal mechanism to function effectively. Results from our study demonstrate that certain aerodynamic and acoustic parameters distinguish adductor spasmodic dysphonia from normal voice.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.15 no.2
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• pp.112-117
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• 2004

Background and Objectives : This study is purposed to investigate the statistically significant discrimination model for predicting vocal fold nodule and polyp's lesional grade, with patients' background data and objective voice evaluation parameters. Materials and Method : The retrospective research was carried out at the Ewha Womans University Hospital. 122 patients' voice examination data had been selected, and lesion screening (Grade I, II, and III) was conducted by 2 ENT specialists, with each patient's vocal fold pictures achieved during the laryngoscopy examination. Results : The Lesional Grade Discrimination Model with which the lesional grade of vocal fold nodules and polyps could be predicted was derived by the ordinal logistic regression analysis (using SPSS 10.0). With this model the lesional grades of 73 out of 122 patients(59.8%) were correctly predicted to their formerly screened ones. Conclusion : This model applied the multivariate approach, which statistically combined these currently used parameters, Jitter, Shimmer, MFR, MPT, and patient's background data such as gender and dysphonia period. It might explain the status of benign lesion of vocal folds, and furthermore expect the physiological function of vocal folds.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.957-963
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• 2007

In this paper, Nonlinear Autoregressive (NAR) method based on Least Square-Support Vector Regression (LS-SVR) is introduced and tested for nonlinear sustained vowel modeling. In the database of total 43 sustained vowel of Benign Vocal Fold Lesions having aperiodic waveform, this nonlinear synthesizer near perfectly reproduced chaotic sustained vowels, and also conserved the naturalness of sound such as jitter, compared to Linear Predictive Coding does not keep these naturalness. However, the results of some phonation are quite different from the original sounds. These results are assumed that single-band model can not afford to control and decompose the high frequency components. Therefore multi-band model with wavelet filterbank is adopted for substituting single band model. As a results, multi-band model results in improved stability. Finally, nonlinear sustained vowel modeling using NAR based on LS-SVR can successfully reconstruct synthesized sounds nearly similar to original voiced sounds.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.446-456
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• 2007

Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.

• The Transactions of the Korea Information Processing Society
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• v.5 no.11
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• pp.2954-2972
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• 1998

It is expected that the transport of compressed video will become a significant part of total network traffic because of the widespread introduction of multimedial services such as VOD(video on demand). Accordingly, VBR(variable bit-rate) encoded video will be widely used, due to its advantages in statistical multiplexing gain and consistent vido quality. Since the transport of video traffic requires larger bandwidth than that of voice and data, the characterization of video source and traffic modeling is very important for the design of proper resource allocation scheme in ATM networks. Suitable statistical source models are also required to analyze performance metrics such as packet loss, delay and jitter. In this paper, we analyzed and described on the characterization and traffic modeling of MPEG video sources. The models are broadly classified into two categories; i.e., statistical models and deterministic models. In statistical models, the models are categorized into five groups: AR(autoregressive), Markov, composite Marko and AR, TES, and selfsimilar models. In deterministic models, the models are categorized into $({\sigma},\;{\rho}$, parameterized model, D-BIND, and Empirical Envelopes models. Each model was analyzed for its characteristics along with corresponding advantages and shortcomings, and we made comparisons on the complexity of each model.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.3
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• pp.9-14
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• 2020

In this paper, Qualification Model of frequency synthesizer is designed for X-band SAR system and performed electrical and environment test. Designed frequency synthesizer generate 13.65 GHz with very low phase noise performance. The integrated phase noise from 10Hz to 1MHz is -37.91 dBc. IRF performances are analyzed according to phase noise and jitter. Also, thermal and structure analysis are achieved for stable operation in space environment. Designed frequency synthesizer is consist of 2 modules of 6U size and generate L-band, C-band, Ku-band. The result of this study would enhance the design ability of RF module and help the frequency synthesizer design for SAR payload system.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.4
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• pp.1927-1943
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• 2016

Jitterbug is a passive network covert timing channel supplying reliable stealthy transmission. It is also the basic manner of some improved covert timing channels designed for higher undetectability. The existing entropy-based detection scheme based on training sample binning may suffer from model mismatching, which results in detection performance deterioration. In this paper, a new detection method based on the feature of Jitterbug covert channel traffic is proposed. A fixed binning strategy without training samples is used to obtain bins distribution feature. Coefficient of variation (CV) is calculated for several sets of selected bins and the weighted mean is used to calculate the final CV value to distinguish Jitterbug from normal traffic. Furthermore, the timing window parameter of Jitterbug is estimated based on the detected traffic. Experimental results show that the proposed detection method can achieve high detection performance even with interference of network jitter, and the parameter estimation method can provide accurate values after accumulating plenty of detected samples.