• Journal of Advanced Navigation Technology
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• v.18 no.1
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• pp.67-73
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• 2014

Due to the development of the communication technology and the high-performance mobile device, the number of video streaming service user has been increasing. Video streaming services in the restricted network environments need error control to support users for the Ultra High Definition(UHD) video delivery. Packet-level FEC method was used to control the error for delay-sensitive video service, but it has a defect in huge occupation of network resources by redundant packets. In this paper, we proposed a packet-level FEC algorithm which maintains the video quality by previous FEC methods and also reduces the occupation of network resource in the restricted network environments. The proposed algorithm controls the redundant rate with using the delivery characteristic of the video coding technology, and improves the efficiency of network resources by reducing the traffic by around 33% in comparison with the previous FEC methods.

• Journal of Communications and Networks
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• v.16 no.5
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• pp.548-558
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• 2014

Load distribution is vital to the performance of multipath transport. The task becomes more challenging in real-time multimedia applications (RTMA), which impose stringent delay requirements. Two key issues to be addressed are: 1) How to minimize end-to-end delay and 2) how to alleviate packet reordering that incurs additional recovery time at the receiver. In this paper, we propose sub-packet based multipath load distribution (SPMLD), a new model that splits traffic at the granularity of sub-packet. Our SPMLD model aims to minimize total packet delay by effectively aggregating multiple parallel paths as a single virtual path. First, we formulate the packet splitting over multiple paths as a constrained optimization problem and derive its solution based on progressive approximation method. Second, in the solution, we analyze queuing delay by introducing D/M/1 model and obtain the expression of dynamic packet splitting ratio for each path. Third, in order to describe SPMLD's scheduling policy, we propose two distributed algorithms respectively implemented in the source and destination nodes. We evaluate the performance of SPMLD through extensive simulations in QualNet using real-time H.264 video streaming. Experimental results demonstrate that: SPMLD outperforms previous flow and packet based load distribution models in terms of video peak signal-to-noise ratio, total packet delay, end-to-end delay, and risk of packet reordering. Besides, SPMLD's extra overhead is tiny compared to the input video streaming.

• Journal of the Korean Society for Industrial and Applied Mathematics
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• v.12 no.3
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• pp.171-190
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• 2008

With growing applications of wireless video streaming, an efficient video traffic model featuring modern high-compression techniques is more desirable than ever, because the wireless channel bandwidths are ever limited and time-varying. We propose a modeling and analysis method for video traffic by a class of stochastic processes, which we call '$GEO^Y/G/{\infty}$ input processes'. We model video traffic by $GEO^Y/G/{\infty}$ input process with gamma-distributed batch sizes Y and Weibull-like autocorrelation function. Using four real-encoded, full-length video traces including action movies, a drama, and an animation, we evaluate our modeling performance against existing model, transformed-M/G/${\infty}$ input process, which is one of most recently proposed video modeling methods in the literature. Our proposed $GEO^Y/G/{\infty}$ model is observed to consistently provide conservative performance predictions, in terms of packet loss ratio, within acceptable error at various traffic loads of interest in practical multimedia streaming systems, while the existing transformed-M/G/${\infty}$ fails. For real-time implementation of our model, we analyze G/D/1/K queueing systems with $GEO^Y/G/{\infty}$ input process to upper estimate the packet loss probabilities.

• Journal of Advanced Navigation Technology
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• v.13 no.4
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• pp.523-531
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• 2009

In this study, we propose a joint source/channel rate control algorithm for video encoder targeting packet erasure channel. Based on the buffer constraints of video communication systems, encoding rate constraint is presented. After defining source distortion models for coded and skipped video frames and a channel distortion model for packet errors and their propagation, an average distortion model of received video is proposed for a given encoding window. Finally, we define an optimization problem to minimize the average distortion for given channel rates and packet loss rates by controlling spatio-temporal parameters of source video and FEC block sizes. Then, we propose a window-based algorithm to solve the problem in real-time.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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• pp.349-350
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• 2009

In this paper, we describe an error concealment techniques based on image inpainting for the image impairments due to the packet loss. Image inpainting is to remove or restore the damaged sections from the images, which is usually old images, paintings, or video films. Inpainting has a long history which goes back to the era when the paintings come out. Manual inpainting is no more used, and we can use digital inpainting for the digitally impaired images and video sequences. In this paper, we review the error concealment techniques for the packet loss recovery and propose our inpainting based image impairment recovery scheme for video communication over packet networks.

• ETRI Journal
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• v.26 no.6
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• pp.535-544
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• 2004

With video compression standards such as MPEG-4, a transmission error happens in a video-packet basis, rather than in a macroblock basis. In this context, we propose a semantic error prioritization method that determines the size of a video packet based on the importance of its contents. A video packet length is made to be short for an important area such as a facial area in order to reduce the possibility of error accumulation. To facilitate the semantic error prioritization, an efficient hardware algorithm for face tracking is proposed. The increase of hardware complexity is minimal because a motion estimation engine is efficiently re-used for face tracking. Experimental results demonstrate that the facial area is well protected with the proposed scheme.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.11
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• pp.673-679
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• 2005

This Paper presents the 3D wavelet transformation based video compression system, which possesses the capability of progressive transmission by increasing resolution and increasing rate for multimedia applications. The 3D wavelet packet based video compression system removes the temporal correlation of the input sequences using the motion compensation filter and decomposes the spatio-temporal subband using the spatial wavelet packet transformation. The proposed system allocates the higher bit rate to the low frequency image of the 3D wavelet sequences and improves the 0.49dB PSNR performance of the reconstructed image in comparison with that of H.263. In addition to the limitation on the propagation of the motion compensation error by the 3D wavelet transformation, the proposed system progressively transmits the input sequence according to the resolution and rate scalability.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.796-806
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• 2009

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.341-357
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• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.12
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• pp.539-546
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• 2002

In this paper, we consider a wireless multimedia environment to service both real-time video traffic and non-real-time WWW-application traffic In our suggested new packet scheduling algorithm, we consider both the accumulation counter and SIR to reduce delay in real-time traffic. In addition, our packet scheduling algorithm gives priority first to real-time video traffic service and then to non-real-time internet Packet service when real-time traffic service is absent. From the simulation results, we find that the AC (Accumulation Counter) scheme has much smaller delay than the conversional M-LWDF scheme for real-time video data users, which has a special quality sensitive to its own delay. We also consider the transmission structure of using both the frame period in the time-axis and the OVSF codes in the code-axis at the same time, which is similar to the structure of HSDPA system.