• Journal of Information Processing Systems
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• v.10 no.2
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• pp.176-192
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• 2014

The realization of Wireless Multimedia Sensor Networks (WMSNs) has been fostered by the availability of low cost and low power CMOS devices. However, the transmission of bulk video data requires adequate bandwidth, which cannot be promised by single path communication on an intrinsically low resourced sensor network. Moreover, the distortion or artifacts in the video data and the adherence to delay threshold adds to the challenge. In this paper, we propose a two stage Quality of Service (QoS) guaranteeing scheme called Prioritized Multipath WMSN (PMW) for transmitting H.264 encoded video. Multipath selection based on QoS metrics is done in the first stage, while the second stage further prioritizes the paths for sending H.264 encoded video frames on the best available path. PMW uses two composite metrics that are comprised of hop-count, path energy, BER, and end-to-end delay. A color-coded assisted network maintenance and failure recovery scheme has also been proposed using (a) smart greedy mode, (b) walking back mode, and (c) path switchover. Moreover, feedback controlled adaptive video encoding can smartly tune the encoding parameters based on the perceived video quality. Computer simulation using OPNET validates that the proposed scheme significantly outperforms the conventional approaches on human eye perception and delay.

• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.121-124
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• 2002

The proxy mechanism widely used in WWW systems offers low-delay data delivery by means of "proxy server". By applying the proxy mechanisms to the video streaming system, we expect that high-quality and low-delay video distribution can be accomplished without introducing extra load on the system. In addition, it is effective to adapt the quality of cached video data appropriately in the proxy if user requests are diverse due to heterogeneity in terms of the available bandwidth, end-system performance, and user′s preferences on the perceived video quality. We have proposed proxy caching mechanisms to accomplish the high-quality and highly-interactive video streaming services. In our proposed system, a video stream is divided into blocks for efficient use of the cache buffer. The proxy server is assumed to be able to adjust the quality of a cached or retrieved video block to the request through video filters. In this paper, to verify the practicality of our mechanisms, we implemented them on a real system and conducted experiments. Through evaluations from several performance aspects, it was shown that our proposed mechanisms can provide users with a low-latency and high-quality video streaming service in a heterogeneous environment.

• ETRI Journal
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• v.38 no.1
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• pp.185-194
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• 2016

In this paper, a novel rate control scheme for low delay video coding of High Efficiency Video Coding (HEVC) is proposed. The proposed scheme is developed by considering a new temporal prediction structure of HEVC. In the proposed scheme, the relationship between bit rate and quantization step is exploited firstly to formulate an accurate quadratic rate-quantization (R-Q) model. Secondly, a method of determining the quantization parameters (QPs) for the first frames within a group of pictures is proposed. Thirdly, an accurate frame-level bit allocation method is proposed for HEVC. Finally, based on the proposed R-Q model and the target bit allocated for the frame, the QPs are predicted for coding tree units by using rate-distortion (R-D) optimization. We compare our scheme against that of three other state-of-the-art rate control schemes. Experimental results show that the proposed rate control scheme can increase the Bjøntegaard delta peak signal-to-noise ratio by 0.65 dB and 0.09 dB on average compared with the JCTVC-I0094 and JCTVC-M0036 schemes, respectively, both of which have been implemented in an HEVC test model encoder; furthermore, the proposed scheme achieves a similar R-D performance to Wang's scheme, as well as obtaining the smallest bit rate mismatch error of all the schemes.

• ETRI Journal
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• v.19 no.4
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• pp.344-362
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• 1997

The delay which is one of the quality of service parameters is considered to be a crucial factor for the effective usage of real-time audio and video streams in interactive multimedia collaborations. Among the various causes of the delay, we focus in this paper on the local delay concerned with the schemes which handle continuous inflow of encoded data from constant or variable bit-rate audio and video encoders. We introduce two kinds of implementation approaches, pull model and push model. While the pull model periodically pumps out the incoming data from the system buffer, the push model receives events from the device drivers. From our experiments based on Windows NT 3.51, it is shown that the push model outperforms the other for both constant and variable bit-rate streams in terms of the local delay, when the system suffers reasonable loads. We interpret this experimental data with M/G/1 multiple vacation queuing theories, and show that it is consistent with the queuing theoretic interpretations.

• Journal of KIISE
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• v.41 no.10
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• pp.824-831
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• 2014

HTTP (Hypertext Transfer Protocol) Adaptive Streaming is gaining attention because it changes bitrates to adapt changing network conditions. Since HAS (HTTP Adaptive Streaming) client downloads the video data based on TCP (Transmission Control Protocol), it estimates incorrectly the available bandwidth and leads to an unnecessary video quality change in long-delay networks. In this paper, we propose a buffer-based quality control scheme in order to improve the service quality and smooth playback in the HAS. The proposed scheme estimates accurately the available bandwidth based on a modified streaming model that considers network delay. It also calculates the sustainability of the video quality to prevent an unnecessary quality change and determines the inter-request time on the basis of the buffer status. Through the simulation, we prove that our scheme improves the QoS (Quality of Service) of the HAS service and controls the video quality smoothly in long-delay networks.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.225-228
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• 2000

In this paper, a design of the reliable video-conference system using CTI(Computer Telephony Integration) technology is proposed. When video-conference is run on the current existing Internet, the transmission delay problem for voice data traffic can be frequently occurred. In order to transmit the real-time voice data through the Internet efficiently, some complicated algorithms such as CODEC(Code/Decode) should be applied. It can cause further excessive processing delay which can affect the overall performance. The voice traffic is usually transmitted through the reliable PSTN(Public Switched Telephone Network) in the CTI system. In this paper a new architecture, in which PSTN for voice traffic and Internet for video traffic are used at the same time instead of using Internet by itself, is proposed to relieve the problems on a video conference.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.798-806
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• 2005

In this paper, we suggest a game system that can support network modules for multi-platform based video games, and built a system that can convert from a single-user game to multi-user game. In this system, we bring in an initial delay buffering scheme on clients to handle any periods of latency occurring from the load fluctuation in a network, when a real-time game is played, and shows that stable play for a game is achieved as the result of the scheme. This paper also presents a retransmission algorithm based on the history of game commands to handle drawbacks of UDP mechanism. And, we evaluate the network delay and packet loss using the simulation tool NS2, and shows the case of 0.3 second buffer delay is the most suitable for recovery.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.5
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.