• Journal of Internet of Things and Convergence
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• v.8 no.1
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• pp.53-58
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• 2022

Smoothing is a transmission plan that converts video data stored at a variable bit rate into a constant bit rate. In the study of [6-7], when a data rate increase is required, the frame with the smallest increase is set as the start frame of the next transmission rate section, when a data tate decrease is required. the frame with the largest decrease is set as the start frame of the next transmission rate section, And the smoothing algorithm was proposed and performance was evaluated in an environment where network traffic is not considered. In this paper, the smoothing algorithm of [6-7] evaluates the adaptive CBA algorithm and performance with minimum frame rate, average frame rate, and frame rate variation from 512KB to 32MB with E.T 90 video data in an environment that considers network traffic. As a result of comparison, the smoothing algorithm of [6-7] showed superiority in the comparison of the minimum refresh rate.

• Journal of IKEEE
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• v.13 no.4
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• pp.29-36
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• 2009

The packet loss and the disconnection during handoff are the most critical problems which degrade the video quality in wireless video streaming. To cope with these problems, we propose an efficient video streaming method in this paper, which does not only dynamically adjust the video transmission rate based on the raptor forward error correction (FEC) level, but also minimize the error propagation during handoff. Firstly, the channel bandwidth of the wireless broadband internet, called WiBro, is estimated by analyzing channel parameters including the carrier to interference and noise ratio (CINR) and the handoff. Secondly, the streaming server adjusts the next transmission rate according to the estimated channel bandwidth and the raptor FEC level to avoid packet error. Also, the encoder performs the intra refresh method that inserts an intra frame (I-frame) right after handoff to reduce the error propagation effectively. Experimental results indicate that the proposed method can improve the performance of the video streaming over WiBro network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.26-38
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• 2006

To construct seamless collaborative environments, what all participants intent should be delivered, and visual elements such gesture, facial expression, and ambiance should be shared with all participants. In this paper, we propose high-quality video service to support DV(digital video) and HDV(high-definition DV) based on Access Grid(AG) which is a prevalent collaborative system. The proposed service is designed for employing versatile media tools and codecs with SDP(session description protocol) and SAP(session announcement protocol). We also design network-adaptive video transmission module to mitigate the impact of network fluctuation. This periodically monitors multicast performance and controls frame rate on sender side considering network condition. The experimental results over the test bed show that proposed service enhances quality of AG video service and provides seamless high-quality video transport by mitigating the impact of network fluctuation.

• Journal of Korea Multimedia Society
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• v.15 no.12
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• pp.1464-1474
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• 2012

In this paper, we propose a QoS (Quality of Service)-aware and cooperative resource reservation scheme using cross-layer link adaptation for wireless high definition video transmission through UWB (Ultra Wide Band) network with D-MAC (Distributed Medium Access Control). A wireless high definition video transmission system usually requires stable high throughput even without line-of-sight, e.g., a destination device in another room separated by a wall. Since the WiMedia D-MAC supporting DRP (Distributed Reservation Protocol) scheme causes lots of DRP resource reservation conflicts due to failure of beacon detection in wireless channel environment, overall performances of the WiMedia D-MAC can be deteriorated. And the current WiMedia MAC standard has not considered QoS provisioning even though QoS parameters such as a range of service rates are provided to each traffic stream. Therefore, we propose Relay DRP protocol with QoS-based relay node selection criterion, which makes a relay path to avoid DRP resource reservation conflicts and guarantee QoS more stably through cross-layer link adaptation of cooperative relay transmission scheme and is compliant with the current WiMedia D-MAC protocol. Simulation results demonstrate performance improvements of the proposed method for throughput and QoS provisioning.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.6
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• pp.341-349
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• 2014

Dynamic adaptive streaming over HTTP (DASH) is the most recent and promising technology to support high quality streaming services. In dynamic adaptive streaming over HTTP (DASH), a client consecutively estimates the available network bandwidth and decides the transmission rate for the forthcoming video chunks to be downloaded. In this paper, we propose a novel rate adaptation algorithm called quality of experience QoE-enhanced adaptation algorithm over DASH (QAAD), which preserves the minimum buffer length to avoid interruption and minimizes the video quality changes during the playback. We implemented a DASH test bed and conducted extensive experiments. Experimental results demonstrate that under fluctuating network conditions, QAAD provides seamless streaming with stabilized video quality while the previous buffer-aware algorithm (i.e., QDASH[9]) frequently changes the video quality and undergoes the interruption.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.105-112
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• 2011

We propose an adaptive load balancing scheme to transport real-time video streams efficiently in this paper. The playback buffer level of a video requesting client is high or low temporarily in mobile environment. This scheme attempts to allocate more network bandwidth to serve a video request with the lower buffer level preferentially. In this scheme, the amount of network bandwidth is dynamically allocated to the requesting clients according to their playback buffer levels in a distributed mobile system. In order to improve the quality of service and real-time performance of individual video playback, the proposed load balancing scheme tries to maximize the number of frames that are transported successfully to the client prior to their playback times. Fair services can also be provided to all the concurrent clients by making their playback situation more adaptive. The performance of this load balancing scheme is compared with that of other static load balancing scheme through extensive simulation experiments, resulting in the higher ratio of frames transmitted successfully within given deadlines.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.1
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• pp.1-10
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• 2006

To reduce the complexity of video transcoder, transcoding techniques generally use intermediate data such as macroblock types, motion vectors, DCT coefficients, which are extracted from incoming video bitstream. In this Paper, we Propose a new adaptive INTRA update scheme utilizing the importance of each macroblock in a given frame. The importance measure is calculated from the motion vectors which are extracted from the decoding procedure of a incomming video data.'rho ratio of the macroblock to be referred by the next frame is calculated by the motion vectors of the next frame and the macroblocks to be refered higher ratio has more effect on the next frame and the followed frames. The proposed scheme can obtain transcoded bitstreams which has better image quality and more improved resilience against transmission errors.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.583-591
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• 2008

This paper proposes a resource allocation algorithm for minimizing the overall distortion of multiple users in orthogonal frequency division multiplex (OFDM). The proposed algorithm exploits the diversity of multiuser and the rate-distortion function using packet distortion model in a system with limited resources. We first induce a rate-distortion function considering error concealment and error propagation properties of H.264 video structures. Then we perform adaptive resource allocation utilizing multiuser diversity for minimizing the overall video quality degradation. We also consider the proportional rate which is pre-determined for each user. Simulation results show that compared to the previous time division multiple access method and the resource allocation method maximizing data rate, the proposed rate allocation algorithm substantially improves the received video quality.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.