• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.8
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• pp.1582-1588
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• 2007

According to recently presented QoS architecture by 3GPP, a traffic conditioner may be deployed to provide conformance of the negotiated QoS. In this paper, a real-time frame-layer rate control method which can be applied to the traffic conditioner of 3GPP is proposed. The proposed rate control method uses an efficient frame skipping algorithm method for low computational complexity, and performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. The proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder.

• Journal of Power Electronics
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• v.19 no.2
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• pp.487-496
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• 2019

For a solid-state transformer (SST), some factors, such as signal delay, switching loss and differences in the system parameters, lead to unbalanced DC-link voltages among the cascaded H-bridges (CHB). With a control method implemented in the ${\alpha}{\beta}$ frame, the DC-link voltages are balanced, and the reactive power is equally distributed among all of the H-bridges. Based on the ${\alpha}{\beta}$ frame control, the system can achieve independent active current and reactive current control. In addition, the control method of the high-voltage stage is easy to implement without decoupling or a phase-locked loop. Furthermore, the method can eliminate additional current delays during transients and get the dynamic response rapidly without an imaginary current component. In order to carry out the controller design, the vector refactoring relations that are used to balance DC-link voltages are derived. Different strategies are discussed and simulated under the unbalanced load condition. Finally, a three-cell CHB rectifier is constructed to conduct further research, and the steady and transient experimental results verify the effectiveness and correctness of the proposed method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.3
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• pp.1123-1140
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• 2019

Many Internet of Things (IoT) applications involving bursty traffic have emerged recently with event detection. A power management scheme qualified for uplink bursty traffic (PM-UBT) is proposed by distinguishing between bursty and general uplink traffic patterns in the IEEE 802.11 standard to balance energy consumption and uplink latency, especially for stations with limited power and constrained buffer size. The proposed PM-UBT allows a station to transmit an uplink bursty frame immediately regardless of the state. Only when the sleep timer expires can the station send uplink general traffic and receive all downlink frames from the access point. The optimization problem (OP) for PM-UBT is power consumption minimization under a constrained buffer size at the station. This OP can be solved effectively by the bisection method, which demonstrates a performance similar to that of exhaustive search but with less computational complexity. Simulation results show that when the frame arrival rate in a station is between 5 and 100 frame/second, PM-UBT can save approximately 5 mW to 30 mW of power compared with an existing power management scheme. Therefore, the proposed power management strategy can be used efficiently for delay-intolerant uplink traffic in event-driven IoT applications, such as health status monitoring and environmental surveillance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12C
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• pp.1208-1215
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• 2006

In this paper, we propose an algorithm that creates three-dimensional (3D) stereoscopic video from two-dimensional (2D) video encoded by H.264 instead of using the conventional stereo-camera process. Motion information of each frame can be obtained by the given motion vectors in most of videos encoded by MPEG standards. Especially, we have accurate motion vectors for H.264 streams because of the availability of a variety of block sizes. 2D/3D video conversion algorithm proposed in this paper can create the left and right images that correspond to the original image by using cut detection method, delay factors, motion types, and image types. We usually have consistent motion type na direction in a given cut because each frame in the same cut has high correlation. We show the improved performance of the proposed algorithm through experimental results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.1A
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• pp.58-64
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• 2009

In this paper, we propose an dynamic random channel allocation scheme increasing probability of channel acquisition and reducing delay by limiting the number of minimum RCHs(Random CHannels) as an optimal factor $\beta$. The MAC protocol of HIPERLAN/2 is based on TDMA/TDD and MT(Mobile Terminal) can obtain chance of traffic transmission through channel competition in RCH period. And AP(Access Point) can dynamically schedule the number of RCHs based on the number of the collision in each RCH of the previous frame. Therefore, the proposed scheme increases a probability of channel acquisition and reduces delay by means of guaranteeing the number of minimum RCHs. With a practical Internet traffic, it is concluded that by the proposed scheme the delay reduction of the order of 19% and slightly better throughput are obtained compared to the conventional algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.5
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• pp.931-936
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• 2016

The PS algorithm divides the number of tags within the identification range of reader into smaller groups by increasing the transmission power incrementally and identifies them. It limits the number of responding tags by grouping the tags within the identification range of the reader, and thus can reduce the probability of tag collision. Also, in the PS algorithm, the reader takes advantages of the difference of identification ranges. This algorithm uses the fixed frame size at every scan. Therefore, it has problems that the performance can be shown variously according to the number of tags and frame size. In this paper, we propose an EPS algorithm that allocates the optimal frame size by estimating the number of tags at each scan, and apply it into the STAC protocol. The simulation results showed that STAC/EPS algorithm can improve the identification delay about 45% compared with STAC protocol. Also, it provides a stable identification delay regardless of power level increase.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.3
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• pp.28-35
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• 2011

Cooperative ARQ scheme is effective for better QoS guarantee for the next generation mobile communication systems where multimedia data transmission highly increases. In this paper, we propose a cooperative ARQ strategy in ad hoc cognitive relays for mobile multimedia communication for supporting instantaneous cooperation in MANET environment. In the proposed strategy, to support real time, delay-sensitive services, whenever a frame is transmitted from the source, each relay actively senses the SINR of the signal transmitted from the source, and determine whether to propose retransmission or not before the destination transmits feedback signal. To minimize the false retransmission decision or needless retransmission, we propose an adaptive sensing threshold optimization algorithm to maintain suboptimal sensing thresholds for each relay. By computer simulation, it is shown that the proposed cooperative ARQ retransmission scheme outperforms the conventional schemes with respect to frame transmission delay and frame loss probability in real time multimedia data transmission system.

• The Transactions of the Korean Institute of Power Electronics
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• v.27 no.4
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• pp.283-290
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• 2022

This study proposes a voltage control scheme in a synchronous reference frame to improve the dynamic characteristics of double-conversion UPSs. UPSs need to control positive and negative sequence voltage, so that positive and negative sequence extractors are generally used to obtain each sequence of the voltage and current. Voltage and current controllers for each sequence are implemented. However, the extractor causes considerable delay, and the delay restricts the control performance, especially for the current controller. To improve the dynamics of the current controller, the proposed scheme adopts a unified current controller without separating positive and negative sequences. By using discrete-time current controller, the control bandwidth can be extended significantly so that negative sequence current can be controlled. To enhance the performance, an additional feed-forward technique for output voltage regulation is proposed. The validity of the proposed controller is verified by experiments.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.