• Journal of Korea Multimedia Society
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• v.20 no.1
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• pp.1-9
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• 2017

Explosive deployment of smart mobile devices such as smart phones, and tablets along with expansion of wireless internet bandwidth have enabled the deployment of mobile video streaming such that video traffic becomes the most important service in wireless networks. Recently, for more efficient video streaming services, the ISO MPEG group standardized a protocol called DASH (Dynamic Adaptive Streaming over HTTP) and the standard has been quickly adopted by many service providers such as YouTube and Netflix. Despite of the convenience of mobile streaming services, users also suffer from low QoE(Quality of Experience) due to dynamic channel fluctuations and unnecessary downloading due to high churning rates. This paper proposes a noble efficient video rate selection algorithm considering user buffer level, channel condition and churning rate. Computer simulation based performance study showed that the proposed algorithm improved the QoE significantly compared to a method that determines the video rate based on current channel conditions. Especially, the proposed method reduced the rebuffering rate, one of the most important performance factors of the QoE, to a nonnegligible level.

• Journal of KIISE:Computer Systems and Theory
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• v.34 no.10
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• pp.529-538
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• 2007

Fair Queueing algorithms based on Generalized Processor Sharing (GPS) not only guarantee sessions with service rate and delay, but also provide sessions with instantaneous fair sharing. This fair sharing distributes server capacity to currently backlogged sessions in proportion to their weights without regard to the amount of service that the sessions received in the past. From a long-term perspective, the instantaneous fair sharing leads to a different quality of service in terms of delay and bandwidth to sessions with the same weight depending on their traffic pattern. To minimize such long-term unfairness, we propose a delay-bandwidth normalization model that defines the concept of value of service (VoS) from the aspect of both delay and bandwidth. A model and a packet-by-packet scheduling algorithm are proposed to realize the VoS concept. Performance comparisons between the proposed algorithm and algorithms based on fair queueing and service curve show that the proposed algorithm provides better long-term fairness among sessions and that is more adaptive to dynamic traffic characteristics without compromising its service rate and delay guarantees.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• Journal of Broadcast Engineering
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• v.17 no.4
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• pp.584-595
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• 2012

Recently, 3D stereoscopic video and HTTP adaptive streaming technologies have received a lot of attention from relevant industrial fields and markets in terms of multimedia contents and delivery services, respectively. It is expected that promising and marketable service models can be created by means of these noticeable two technologies. However, current standard specifications do not provide a method for organized connection between those two technologies. 3D stereoscopic video services are weighted in broadcasting and storage services that are only available under environments in which the network bandwidth is guaranteed or free. Also, HTTP adaptive streaming technologies only provide plain 3D service methods that are dependent on particular Codec. Therefore, this paper proposes 3D video delivery format for HTTP adaptive streaming service which enables stable and seamless display for various stereoscopic video sequences over internet networks. The proposed technology is designed on the basis of Stereoscopic Video Application Format which is a service-oriented standard specification for storing stereoscopic video sequences. Also, this delivery format is directly applicable over DASH that is the representative standard technology for HTTP adaptive streaming services. The delivery format proposed in this paper has been submitted to MPEG and it has been accepted as a working draft, thus it expected to pave the way for practical industrialization in relevant fields from now on.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.8
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• pp.2796-2813
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• 2014

This paper presents a new algorithm called the adaptive logarithmic increase and adaptive decrease algorithm (A-LIAD), which mainly addresses the Round-Trip Time (RTT) fairness problem in satellite networks with a very high propagation delay as an alternative to the current TCP congestion control algorithm. We defined a new increasing function in the fashion of a logarithm depending on the increasing factor ${\alpha}$, which is different from the other logarithmic increase algorithm adopting a fixed value of ${\alpha}$ = 2 leading to a binary increase. In A-LIAD, the ${\alpha}$ value is derived in the RTT function through the analysis. With the modification of the increasing function applied for the congestion avoidance phase, a hybrid scheme is also presented for the slow start phase. From this hybrid scheme, we can avoid an overshooting problem during a slow start phase even without a SACK option. To verify the feasibility of the algorithm for deployment in a high-speed and long-distance network, several aspects are evaluated through an NS-2 simulation. We performed simulations for intra- and interfairness as well as utilization in different conditions of varying RTT, bandwidth, and PER. From these simulations, we showed that although A-LIAD is not the best in all aspects, it provides a competitive performance in almost all aspects, especially in the start-up and packet loss impact, and thus can be an alternative TCP congestion control algorithm for high BDP networks including a satellite network.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.11
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• pp.1-8
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• 2007

A wideband continuous-time sigma delta modulator for wireless application is implemented in 130nm CMOS. The SNR for small input signal is improved using a proposed adaptive quantizer which can effectively scale the quantization level. The modulator comprises a second-order loop filter for low power consumption, 4-bit quantizer and DAC for low jitter sensitivity and high linearity. Designed circuit achieves peak SNR of 51.36B with 10MHz signal Bandwidth and 320MHz sampling frequency dissipating 30mW.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.115-120
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• 2011

This paper is concerned with the performance comparison of CR-CMA (Coordinate Reduction-CMA) and CM-CMA (Constellation Matching-Constant Modulus Algorithm) that is used for improving the convergence characteristic and residual intersymbol interference which are used as the performance index for an adaptive equalizer. The equalizer is used to reduce the distortion caused by the intersymbol interference on the wireless and the wired band-limited channel, and the blind method which does not need for extra bandwidth by the training sequence of digital code are researched. Recently, by using the merit of simple operation in the CMA, the performance improvement is obtained by the modifying the cost function of it. In this paper, the new algorithm, CR-CMA and CM-CMA, the performance analysis are performed and compared by computer simulation. The CR-CMA has a superior equalization characteristics in the recovered constellation, convergence speed and residual intersymbol interference than the CM-CMA by computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.48 no.1
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• pp.31-38
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• 2011

With the increasing demand for high-speed transmission systems, adaptive equalizers have been widely used in receivers to overcome the limited bandwidth of channels. In order to reduce the cost for testing high-speed receiver chips, on-chip eye-opening monitoring (EOM) technique which measures the eye-opening of data waveform inside the chip can be employed. In this paper, a 10-Gb/s adaptive 2-tap look-ahead decision feedback equalizer (DFE) with EOM function is proposed. The proposed EOM circuit can be applied to look-ahead DFEs while existing EOM techniques cannot. The magnitudes of the post-cursors are measured by monitoring the eye of received signal, and coefficients of DFE are calculated using them by proposed adaptation algorithm. The circuit designed in 90nm CMOS technology and the algorithm are verified with post-layout simulation. The DFE core occupies $110{\times}95{\mu}m^2$ and consumes 11mW in 1.2V supply voltage.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.62-68
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• 2002

It is well known that the MDVR beamforming outperforms the conventional delay-sum beamformer in the sense of noise rejection and bearing resolution. However, the MDVR method requires long observation time to achieve high frequency resolution. The STMV method uses the steered covariance matrix of sensor data, so it has an ability to form an adaptive weight vector from a single time-series snapshot. But it uses the same weight vector across all frequencies. In this paper, we propose an SSMV method. The basic idea of the SSMV method is to decompose a full frequency band into several subbands to acquire a weight vector for each subband, individually. Also the wrap may be divided into several subarrays in order to reduce a computational load and the bandwidth of each subband. Simulations using real sea trial data show that the proposed SSMV method has good performance with short observation time.