• Journal of Broadcast Engineering
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• v.3 no.1
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• pp.69-84
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• 1998

In digital broadcasting, services such as digital satellite TV, cable TV and digital terrestrial TV, several video programs are compressed by MPEG-2, and then simultaneously transmitted over a conventional CBR (Constant Bit Rate) broadcasting channel. In this paper, we propose a joint quality control scheme to be able to accurately control the relative picture quality among the video programs, which is achieved by simdt;,nL'Ously controlling the video encoders to generate the VBR (Variable Bit Rate) compressed video streams. Our quality control scheme can prevent the video buffer overflow and underflow by total target bit allocation process, and also exactly control the relative picture quality in terms of PSNR (Peak Signal to Noise Ratio) between some programs requiring higher picture quality and others by rate-distortion modification. Furthermore we present a rate-distortion estimation method for MPEG-2 video, which is base of our joint quality control, and verify its performance by experiments. The most attractive features of this estimation method are as follows: 1) computational complexity is low because main operation for the estimation is to calculate the histogram of OCT coefficients into quantizer; 2) estimation results are very accurate enough to be applied to the practical MPEG-2 video coding applications. Simulation results show that the proposed joint quality control scheme accurately controls the relative picture quality among the video progran1s transmitted over a single channel as well as provides more consistent and higher picture quality than independent coding scheme that encodes each program independently.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.8
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• pp.2851-2865
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• 2014

To improve the rate-distortion performance of distributed video compressive sensing (DVCS), the adaptive sparse basis and nonlocal similarity of video are proposed to jointly reconstruct the video signal in this paper. Due to the lack of motion information between frames and the appearance of some noises in the reference frames, the sparse dictionary, which is constructed using the examples directly extracted from the reference frames, has already not better obtained the sparse representation of the interpolated block. This paper proposes a method to construct the sparse dictionary. Firstly, the example-based data matrix is constructed by using the motion information between frames, and then the principle components analysis (PCA) is used to compute some significant principle components of data matrix. Finally, the sparse dictionary is constructed by these significant principle components. The merit of the proposed sparse dictionary is that it can not only adaptively change in terms of the spatial-temporal characteristics, but also has ability to suppress noises. Besides, considering that the sparse priors cannot preserve the edges and textures of video frames well, the nonlocal similarity regularization term has also been introduced into reconstruction model. Experimental results show that the proposed algorithm can improve the objective and subjective quality of video frame, and achieve the better rate-distortion performance of DVCS system at the cost of a certain computational complexity.

• Journal of Broadcast Engineering
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• v.20 no.6
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• pp.921-927
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• 2015

In this paper, we propose a novel design scheme for the operation of Decoded Picture Buffer (DPB) including reference picture re-ordering, marking process, and reference picture list construction to perform an efficient scalable multi-view video coding. Extensive simulations show that the proposed method can provide improved compression efficiency and improved video quality measured in terms of BD-Rate and BD-PSNR for the scalable multi-view video coding.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.12 no.2
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• pp.38-44
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• 2013

In this paper, we designed and implemented a mobile video transmission device with 1080i resolution and 30fps frame rate using wireless multi-access technologies. The video acquisition delay time of 16us and jitter of 21ms for this equipment were obtained, respectively. The device is designed to access various communication networks. In order to evaluate the performance of its video transmission capability, a server and monitoring viewer are implemented. In the performance test using LTE network, 211ms delay time was obtained at 3Mbps transmission. Rendering of 1fps frame rate at the 265kbps transmission was confirmed for the transmission test using Inmarsat network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6C
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• pp.615-623
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• 2003

This paper presents effective quality-of-service renegotiating schemes for streaming video. The conventional network supporting quality-of-service generally allows a negotiation at call setup. However, it is not efficient for the video application since the compressed video traffic is statistically non-stationary. Thus, we consider the network supporting quality-of-service renegotiations during the data transmission, and study effective quality-of-service renegotiating schemes for streaming video. Simple token bucket model, whose parameters are token filling rate and token bucket size, is adopted for the video traffic model. The renegotiating time instants and the parameters are determined by analyzing the statistical information of compressed video traffic. In this paper, two renegotiating approaches, i.e. fixed renegotiating interval case and variable renegotiating interval case, are examined. Finally, the experimental results are provided to show the performance of the proposed schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.11
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• pp.4585-4603
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• 2015

Video streaming services make up a large proportion of Internet traffic on both fixed and mobile access throughout the world. Adaptive streaming allows for dynamical adaptation of the bitrate with varying network conditions, to guarantee the best user experience. Adaptive bitrate algorithms face a significant challenge in correctly estimating the throughput as it varies widely over time. In this paper, we first evaluate the throughput estimation techniques and show that the method that we have used offers stable response to throughput fluctuations while maintaining a stable playback buffer. Then, we propose an adaptive bitrate scheme that intelligently selects the video bitrates based on the estimated throughput and buffer occupancy. We show that the proposed scheme improves viewing experience by achieving a high video rate without taking unnecessary risks and by minimizing the frequency of changes in the video quality. Furthermore, we show that it offers a stable response to short-term fluctuations and responds swiftly to large fluctuations. We evaluate our algorithm for both constant bitrate (CBR) and variable bitrate (VBR) video content by taking into account the segment sizes and show that it significantly improves the quality of video streaming.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.9
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• pp.4572-4586
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• 2019

Mobile Edge Computing (MEC) and Information-Centric Networking (ICN) are essential network architectures for the future Internet. The advantages of MEC and ICN such as computation and storage capabilities at the edge of the network, in-network caching and named-data communication paradigm can greatly improve the quality of video streaming applications. However, the packet loss in wireless network environments still affects the video streaming performance and the existing loss recovery approaches in ICN does not exploit the capabilities of MEC. This paper proposes a Deep Learning based Loss Recovery Mechanism (DL-LRM) for video streaming over MEC based ICN. Different with existing approaches, the Forward Error Correction (FEC) packets are generated at the edge of the network, which dramatically reduces the workload of core network and backhaul. By monitoring network states, our proposed DL-LRM controls the FEC request rate by deep reinforcement learning algorithm. Considering the characteristics of video streaming and MEC, in this paper we develop content caching detection and fast retransmission algorithm to effectively utilize resources of MEC. Experimental results demonstrate that the DL-LRM is able to adaptively adjust and control the FEC request rate and achieve better video quality than the existing approaches.

• Journal of Korea Multimedia Society
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• v.24 no.12
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• pp.1641-1652
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• 2021

To analyze the crime scene, the role of digital evidence such as CCTV and black box is very important. Such digital evidence is often damaged due to device defects or intentional deletion. In this case, the deleted video can be restored by well-known techniques like the frame-based recovery method. Especially, the data such as the video can be generally fragmented and saved in the case of the memory used almost fully. If the fragmented video were recovered in units of images, the sequence of the recovered images may not be continuous. In this paper, we proposed a new video restoration method to match the sequence of recovered images. First, the images are recovered through a frame-based recovery technique. Then, after analyzing the time information marked on the images, the time information was extracted and recognized via optical character recognition (OCR). Finally, the recovered images are rearranged based on the time information obtained by OCR. For performance evaluation, we evaluate the recovery rate of our proposed video restoration method. As a result, it was shown that the recovery rate for the fragmented video was recovered from a minimum of about 47% to a maximum of 98%.

• Information and Communications Magazine
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• v.11 no.8
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• pp.60-77
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• 1994

This paper presents a survey of video coding schemes which use wavelet transform for the videophone on very low bit rate commun ication chan nel( ego 10 Kbps Public Service Telephone Network). Firstly, we introduce the standardization efforts to make the low bit rate videophone architecture and the typical application of low bit rate video coding scheme. Secondly, we summarize the several requirements on videophone, delay, encoder/decoder complexity, low bitrate, and progressive transmission capability. Third, we review the basic theory of wavelet transform without much mathematics. We compare the wavelet transform with short-time fourier transform and subband filters. Fourth, we summarize the video coding schemes proposed so far, and evaluate them with Ule requirements. Lastly, we conclude with fu¬ture research directions.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.4
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• pp.784-799
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• 2013

This paper presents an adaptive rate control scheme for streaming-based content sharing service. This scheme delivers multimedia contents from a user device to another device or seamlessly redirects streaming service across heterogeneous user devices. In the proposed scheme, a streaming server adjusts video quality level according to the network and client status. Our scheme is different from other rate control schemes, because the video quality at the server is decided not only based on the available bandwidth, but also based on the device characteristics and bandwidth requirement at the access network. We also propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows competing for the same narrow link with different Round Trip Times (RTTs). Through the simulation, we prove that our scheme improves the network stability and the quality of streaming service by appropriately adjusting the quality of the video stream. The simulation results also demonstrate the ability of the proposed scheme in ensuring RTT-fairness while remaining throughput efficient.