• 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.8B
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• pp.751-759
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• 2009

Due to the advance of WLAN technology, the use of the multimedia service such as the video streaming service has been increased in the home network. However, we need to study the method which decreases the transmission delay and the frame loss rate to provide QoS of the video streaming service. Therefore, this paper proposes an active buffer management mechanism to guarantee QoS of the streaming service in IEEE 802.11e EDCA. The proposed protocol discards the frame in the HoL of the buffer based on the importance of each frame and the virtual transmission delay of frame newly arriving at the buffer. In the simulation results, the proposed algorithm not only decreases the frame loss probability of important I and P frames but also stabilizes the transmission delay. It may increase the QoS of video streaming services.

• The KIPS Transactions:PartB
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• v.11B no.2
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• pp.177-186
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• 2004

It is a weak point of the motion estimation technique for video compression that the predicted video encoding algorithm requires higher-order computational complexity. To reduce the computational complexity of encoding algorithms, researchers introduced techniques such as 3D-WT that don't require motion prediction. One of the weakest points of previous 3D-WT studies is that they require too much memory for encoding and too long delay for decoding. In this paper, we propose a technique called `FS (Fast playable and Scalable) 3D-WT' This technique uses a modified Haar wavelet transform algorithm and employs improved encoding algorithm for lower memory and shorter delay requirement. We have executed some tests to compare performance of FS 3D-WT and 3D-V. FS 3D-WT has exhibited the same high compression rate and the same short processing delay as 3D-V has.

• Journal of Broadcast Engineering
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• v.10 no.1 s.26
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• pp.83-102
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• 2005

The problem of video scheduling is analyzed in the framework of divisible load scheduling. A divisible load can be divided into any number of fractions (parts) and can be processed/computed independently on the processors in a distributed computing system/network, as there are no precedence relationships. In the video scheduling, a frame can be split into any number of fractions (tiles) and can be processed independently on the processors in the network, and then the results are collected to recompose the single processed frame. The divisible load arrives at one of the processors in the network (root processor) and the results of the computation are collected and stored in the same processor. In this problem communication delay plays an important role. Communication delay is the time to send/distribute the load fractions to other processors in the network. and the time to collect the results of computation from other processors by the root processors. The objective in this scheduling problem is that of obtaining the load fractions assigned to each processor in the network such that the processing time of the entire load is a minimum. We derive closed-form expression for the processing time by taking Into consideration the communication delay in the load distribution process and the communication delay In the result collection process. Using this closed-form expression, we also obtain the optimal number of processors that are required to solve this scheduling problem. This scheduling problem is formulated as a linear pro-gramming problem and its solution using neural network is also presented. Numerical examples are presented for ease of understanding.

• KIPS Transactions on Computer and Communication Systems
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• v.6 no.6
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• pp.255-260
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• 2017

The limitation of real-time video conferencing system is that the delay of network and buffering and the transmission of user information are not efficiently performed between systems, so real - time performance is not guaranteed completely. In order to overcome this problem, the study on the extension of the network infrastructure and the jitter delay is actively carried out, but the study on the buffering delay is insufficient. In this paper, we propose a frame-rate control buffer management (FRCB) scheme to solve the problem caused by buffering delay. The FRCB is used to prevent overflow and underflow of the buffer by adopting the two-stage buffer threshold of Fast-play THreshold (FTH) and Slow-play THreshold (STH). Therefore, it showed better performance than jitter buffer even under high CPU load, and showed that it is suitable for high quality real time video conferencing.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.9
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• pp.2268-2283
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• 2013

Scalable video coding can handle peer heterogeneity of P2P streaming applications, but there is still a lack of comprehensive studies on how to use it to improve video playback quality. In this paper we propose a capacity aware scalable video coding mechanism for P2P on demand streaming system. The proposed mechanism includes capacity based neighbor selection, adaptive data scheduling and streaming layer adjustment, and can enable each peer to select appropriate streaming layers and acquire streaming chunks with proper sequence, along with choosing specific peers to provide them. Simulation results show that the presented mechanism can decrease the system's startup and playback delay, and increase the video playback quality as well as playback continuity, and thus it provides a better quality of experience for users.

• ETRI Journal
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• v.36 no.2
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• pp.253-263
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• 2014

In the near future, traditional narrow and fixed viewpoint video services will be replaced by high-quality panorama video services. This paper proposes a visual-attention-aware progressive region of interest (RoI) trick mode streaming service (VA-PRTS) that prioritizes video data to transmit according to the visual attention and transmits prioritized video data progressively. VA-PRTS enables the receiver to speed up the time to display without degrading the perceptual quality. For the proposed VA-PRTS, this paper defines a cutoff visual attention metric algorithm to determine the quality of the encoded video slice based on the capability of visual attention and the progressive streaming method based on the priority of RoI video data. Compared to conventional methods, VA-PRTS increases the bitrate saving by over 57% and decreases the interactive delay by over 66%, while maintaining a level of perceptual video quality. The experiment results show that the proposed VA-PRTS improves the quality of the viewer experience for interactive panoramic video streaming services. The development results show that the VA-PRTS has highly practical real-field feasibility.

• Journal of Internet Computing and Services
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• v.20 no.1
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• pp.47-55
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• 2019

Free-viewpoint video (FVV) service provides multi viewpoints of contents and synthesizes intermediate video files which are not captured on some view angles so that enables users to watch as they choose wherever they want. Synthesizing video is necessary technique to provide FVV video service, because every video of the FVV contents for different view angles cannot be stored to the content server physically. For the reason, fast view synthesis can improve the quality of video service and increase user's satisfaction. One of the studies for FVV service, a method was proposed to transmit FVV service based on DASH (Dynamic Adaptive Streaming over HTTP). There is big advantage on using DASH that it is commonly used to transport video service. However, the method was only a conceptual proposal, so it is difficult to implement the system using the proposal. In this paper, we propose an implementation method to provide real-time FVV service smoothly. We suggest a system structure and operation method on the server and client side in detail, which is to be applicable to synthesize video quickly. Also, we suggest generating FVV service map additionally which controls a FVV service overall. We manage real-time information of the whole service through the service map. The service can be controlled by reducing the possible delay from network situation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• Journal of Korean Institute of Industrial Engineers
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• v.23 no.1
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• pp.23-37
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• 1997

In this paper, performance analysis of video traffic shaper for Motion Picture Experts Group (MPEG) video traffic in on ATM network are investigated. Traffic shaping for MPEG video traffic is proposed as a traffic control function in ATM networks. The proposed shaper smoothes video traffics by controling the output rate of the buffer, which is placed in an MPEG source, according to I,B,P frame sequences of MPEG. In performance analysis of an video traffic shaper, a periodic botch arrival model is suggested to describe cell streams in a frame of MPEG video traffic. The queueing model which has periodic independent botch arrival and periodic deterministic service time is used to obtain the cell loss ratio, the mean cell delay, and the measure of smoothing effect. Simulation results are used to validate this queueing model. The cell loss performance of ATM multiplexer is measured by simulation study with real MPEG-1 data. From the viewpoint of traffic load, the cell loss ratio is observed to be considerably high, which is considered to result from the burstiness of MPEG video traffic. As a result, it is shown that the shaping decreases cell loss ratio of multiplexer. The results of this paper can be employed to establish a basic guideline in the implementation of a traffic control scheme and the design of ATM multiplexer for MPEG video traffic.