• Journal of KIISE
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• v.45 no.2
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• pp.175-186
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• 2018

Recently, HAS (HTTP Adaptive Streaming) has been the subject of much attention to improve the QoE (Quality of Experience). In a high latency network, HAS degrades the QoE due to the lost RTT cycle since it replies with a response of one segment to the request of one segment. The server-push based HAS schemes of downloading multiple segments in one request cause QoE degradation due to the buffer underflow. In this paper, we propose a VSSDS (Video Streaming Scheme based on Dynamic Server-push) scheme to improve the QoE in a high latency network. The proposed scheme adjust video quality by estimating available bandwidth and determine the number of segments to be downloaded for each segment request cycle. Through the simulation, the proposed scheme not only improves the average video bitrate but also alleviates the buffer underflow.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.275-277
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• 2000

인터넷상에서 화상전화를 구현하고자 할 때는 가변적인 네트웍 대역폭에 따라 전송률을 동적으로 조절하는 대역폭 제어기법이 요구된다. 그러나, 기존의 기법들은 종점간 할당된 대역폭을 미디어의 특성에 상관없이 동일하게 조절함으로써 각 미디어에 맞는 QoS 수준을 충분히 제공하지 못하고 있다. 본 논문에서는 미디어의 특성을 고려하여 각 미디어별로 차별적인 QoS를 제공할 수 있는 미디어간 대역폭 조절 기법과 전송 기법을 제안하고 이를 실험한다. 실험을 위해 PC상에서 화상전화의 프로토타입을 구현하였으며, 실험 결과, 비디오의 전송품질을 크게 영향을 미치지 않고, 오디오의 전송품질을 향상시킬 수 있었다.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.

• KIISE Transactions on Computing Practices
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• v.20 no.11
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• pp.604-609
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• 2014

Decoding processes in portable media players have a high computational cost, resulting in high power consumption by the CPU. If decoding computations are reduced, the power consumed by the CPU is also be reduced, but such a choice generally results in a degradation of the video quality for the users, so it is essential to address this tradeoff. We proposed a new CPU power management scheme that can make use of the scalability property available in the H.164/SVC standard. We first proposed a new video quality model that makes use of a video quality metric(VQM) in order to efficiently take into account the different quantization factors in the SVC. We then propose a new dynamic voltage scaling(DVS) scheme that can selectively combine the previous decoding times and frame sizes in order to accurately predict the next decoding time. We then implemented a scheme on a commercial smartphone and performed a user test in order to examine how users react to the VQM difference. Real measurements show that the proposed scheme uses up to 34% fewer energy than the Linux DVFS governor, and user tests confirm that the degradation in the quality is quite tolerable.

• Broadcasting and Media Magazine
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• v.12 no.4
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• pp.28-39
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• 2007

다양한 디지털 기술의 발전으로 인하여 방송형태의 이동 멀티미디어 서비스가 다국적으로 제안되고, 국내에서는 이동 멀티미디어 방송 (DMB: Digital Multimedia Broadcasting)을 통하여 야외나 이동시에도 시청이 가능한 방송서비스가 활발해지고 있다. 휴대 및 이동수신 방송 환경에서 비디온 오디오 및 데이터를 포함한 멀티미디어 방송 서비스를 효율적으로 제공하기 위해서는 다양한 장소에서 수신 영상에 대한 품질 확보가 필수적이다. 본 논문에서는 현재 이동 멀티미디어 방송이 비디오 압축방식으로 채택하고 있는 H.264/AVC 압축 파라미터의 성능 연구에 대하여 기술한다. 현재 국내의 위성/지상파 DMB의 경우 비디오의 압축 방법으로 H.264/AVC baseline 1.3의 표준규격을 사용한다. 이러한 비디오 코덱(codec) 이용하여 비디오 영상을 압축할 경우 관련 파라미터(parameter) 조절이 가능한데, 비디오를 압축할 경우 관련 파라미터들을 어떻게 정하느냐에 따라 서로 다른 수신환경에서 압축 효율 및 재생된 비디오의 화질에 많은 영향을 미친다. 따라서 수신 환경에 가장 적합한 비디오 화질을 얻기 위해서는 관련 파라미터 설정이 매우 중요하다. 본 논문에서는 다양한 압축 파라미터들 중 화질에 많은 영향을 미치는 항목을 선정하여, 해당 파라미터의 변화가 재생된 비디오 화질에 미치는 영향을 객관적 평가척도인 PSNR, Bit-rate, 수행시간 등을 이용하여 분석하였다. 또한, 실험 결과를 바탕으로 이동 멀티미디어 방송 환경에서의 H.264 인코더의 적정 압축 파라미터 및 인코더의 성능 개선 방안을 제안한다.

• Journal of KIISE
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• v.45 no.1
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• pp.86-93
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• 2018

When multiple clients share bandwidth and receive a streaming service, HTTP Adaptive Streaming has a problem in that the bandwidth is measured inaccurately due to the ON-OFF pattern of the segment request. To solve the problem caused by the ON-OFF pattern, the proposed PANDA (Probe AND Adapt) determines the quality of the segment to be requested while increasing the target bandwidth. However, since the target bandwidth is increased by a fixed amount, there is a problem in low bandwidth utilization and a slow response to changes in bandwidth. In this paper, we propose a video quality control scheme that improves the low bandwidth utilization and slow responsiveness of PANDA. The proposed scheme adjusts the amount of increase in the target bandwidth according to the bandwidth utilization after judging the bandwidth utilization by comparing the segment download time and the request interval. Experimental results show that the proposed scheme can fully utilize the bandwidth and can quickly respond to changes in bandwidth.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2315-2320
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• 2006

A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for low bit rate video coding. The proposed rate control method uses a non-iterative optimization 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. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performances than the existing rate control method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2B
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• pp.334-344
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• 2010

Recently, due to the prevalence of various mobile devices and broadband wireless networks, a significant interests and demands for multimedia streaming services over the Internet have been increasing. However, it is difficult to transmit continuous multimedia stream when mobile terminals are moving. Therefore, in order to deploy mobile IPTV service in the broadband wireless network, efficient wireless resource utilization and seamless QoE (Quality of Experience) offers to the users are an important issue. In this paper, we propose a network based adaptive streaming scheme, called MARC (Mobile Adaptive Rate Control), which controls the quality of the video and rate of the video based on the status of the wireless channel. The proposed scheme uses awareness information of the wireless channel status and controls transmitting streaming video which is suitable for the wireless channel status and mobile station location, in order to provide a seamless video playback for mobile environment in addition to improving the quality of a streaming service. The proposed MARC scheme alleviates the discontinuity of video playback and allocates suitable client buffer in broadband wireless network. Simulation results demonstrate the effectiveness of our proposed scheme.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.2
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• pp.181-185
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• 2008

Layered video multicast such as RLM Receiver-driven layered multicast) is a premising technique for delivering streaming video to a set of heterogeneous receivers over ALM(Application Layer Multicast) as well as over IP multicast. However, this approach may suffer from unnecessary fluctuation of video quality due to overlapped and failed join-experiments. Though a shared teaming scheme was introduced to resolve these problems, it may cause high control overhead and slow convergence problem when used with ALM. In this paper, we propose a new shared learning scheme for ALM-based layered video multicast which reduces control overhead and convergence latency while keeping the number of fluctuation reasonably small. The simulation results show that the proposed scheme performs better than an ALM-based layered video multicast with shared learning in terms of control overhead and convergence latency.

• Journal of KIISE
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• v.43 no.3
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• pp.327-335
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• 2016

In this paper, we propose a buffer management scheme suitable for interactive multimedia services. We consider a typical delay optimization environment so that receiver buffer lengths vary according to the round trip time estimation. In this environment, we propose an optimization technique for minimizing the loss of information that may occur when a reduced buffer length forces I/P/B frames in the buffer to drop. We modeled our problem as a Knapsack Problem for which we used dynamic programing in order to find an approximate solution. The proposed technique is compared with the existing buffer management techniques. Through simulation studies, we found that our approach could increase PSNR, which is important to video quality.