• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.11
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• pp.5271-5289
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• 2016

On large-scale data analysis platforms deployed on cloud infrastructures over the Internet, the instability of the data transfer time and the dynamics of the processing rate require a more sophisticated data distribution scheme which maximizes parallel efficiency by achieving the balanced load among participated computing elements and by eliminating the idle time of each computing element. In particular, under the constraints that have the real-time and limited data buffer (in-memory storage) are given, it needs more controllable mechanism to prevent both the overflow and the underflow of the finite buffer. In this paper, we propose an auto regulated data provisioning model based on receiver-driven data pull model. On this model, we provide a synchronized data replenishment mechanism that implicitly avoids the data buffer overflow as well as explicitly regulates the data buffer underflow by adequately adjusting the buffer resilience. To estimate the optimal size of buffer resilience, we exploits an adaptive buffer resilience control scheme that minimizes both data buffer space and idle time of the processing elements based on directly measured sample path analysis. The simulation results show that the proposed scheme provides allowable approximation compared to the numerical results. Also, it is suitably efficient to apply for such a dynamic environment that cannot postulate the stochastic characteristic for the data transfer time, the data processing rate, or even an environment where the fluctuation of the both is presented.

• The KIPS Transactions:PartB
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• v.11B no.6
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• pp.625-632
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• 2004

Since the H.264 encoding adopts both forward prediction and hi-direction prediction modes and exploits Variable Length Coding(VLC), the amount of data generated from video encoder varies as Flaying time goes by. The fixed bit rate encoding system which has limited transmission channel capacity uses a buffer to control output bitstream It's necessary to control the bitstream to maintain within manageable range so as to protect buffer from overflow or underflow. With existing bit amount control algorithms, the $\lambda_{MODE}$ which is relationship between distortion value and quantization parameter often excesses normal value to end up with video error. This paper proposes an algorithm to protect buffer from overflow or underflow by introducing a new quantization parameter against distortion value of H.264 video data. The test results of 6 exemplary data show that the proposed algorithm has the same PSNR as and up to 8% reduced bit rate against existing algorithms.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.527-539
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• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• Journal of Digital Contents Society
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• v.17 no.3
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• pp.135-142
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• 2016

When video service is performed in simulation using NS-2(Network Simulation-2), the video replay is performed as the received frame order. In the existing video replay method based on the received frame order, as the frame orders of receiver and transmitter are different, the receiver buffer does not have the effect that the packets between the frames of transmitter buffer holds a regular size and packet dense and sparsity phenomenon in the receiver buffer is made by the irregular packet size due to the unpredictable reversed order of received partial frames. The above dense and sparsity phenomenon increases the probability of buffer overflow and underflow generation. To prevent these problems, the proposed frame receive order relocation method adds an extra replay buffer which rearranges the order of receive frame as the order of transmit frame, so it has the effect that the packets between the transmit frames keeps a regular size. Through the simulation using NS-2 and JSVM(Joint Scalable Video Model), the generation number of buffer overflow and underflow, and PSNR(Required Peak Signal to Noise Ratio) performance between the existing method and proposed method were compared. As a result, it was found that the proposed method would have better performance than the existing method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1976-1985
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• 2015

The amount of packets in palyout buffer of video streaming receiver can be changed by network condition, and saturated and exhausted by the delay and jitter. Especially, if the amount of incoming video traffic exceeds the maximum allowed playout buffer, buffer overflow problem can be generated. It makes the deterioration of video image and the discontinuity of playout by skip phenomenon. Also, if the incoming packets are delayed by network confusion, the stop phenomenon of video image is made by buffering due to buffer underflow problem. To solve these problems, this paper proposes the video streaming receiver with token bucket scheme which automatically establishes the important parameters like token generation rate r and bucket maximum capacity c adapting to the pattern of video packets. The simulation results using network simulator-2 (NS-2) and joint scalable video model (JSVM) show that the proposed token bucket scheme with automatic establishment parameter provides better performance than the existing token bucket scheme with manual establishment parameter in terms of the generation number of overflow and underflow, packer loss rate, and peak signal to noise ratio (PSNR) in three test video sequences.

• KIISE Transactions on Computing Practices
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• v.21 no.12
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• pp.780-785
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• 2015

Recently HTTP (Hypertext Transfer Protocol) adaptive streaming services have been the subject of much attention. The video quality control scheme of conventional HTTP adaptive streaming services estimates bandwidth using segment throughput and smooths out the sample of segment throughput. However, the conventional scheme has the problem of QoE (Quality of experience) degradation occurring with buffer underflow and frequent quality change due to the fixed number of samples. In order to solve this problem, we propose a video quality control scheme based on segment throughput and buffer occupancy. The proposed scheme determines the number of samples according to the variation of segment throughput. The proposed scheme also controls video quality based on the threshold of bitrate to keep stable buffer occupancy. The simulation results show that proposed scheme improves QoE by preventing buffer underflow and decreasing quality change when compared with the conventional scheme.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1987.04a
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• pp.31-34
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• 1987

A new efficientcoding system which can transmit video conferenceof viedeophone signals at a 64kbps is proposed. In addition to the interframe and CRC (Conditional Repleni shment Coding) system, BTVQ (Binary Tree searched Vector Quantizer)and RLC (Run Length Coding) methods are incorporated. Couble buffer memory is used for simple comtrol of channel symbol transmission and memory underflow And also buffer memory onerfolw is easily controlled by the thresholds of a MAD (Moving Area Betector)

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.99-102
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• 1998

As multimedia communication services have been widely spreading, the amount of video traffic is rapidly increasing in B-ISDN environment based on the ATM technology. The image quality of MPEG services is very sensitive to the cell losses in ATM network, since each cell contains information needed at decoding process. Since the number of cells in each frame of MPEG is variable, this video smoothing technology need to prepare a buffer for no overflow or underflow at the transmission, requires that some number of cells be taken to the buffer in client before the playback of video. To ensure the high quality image of video, the video smoothing is scheduled by a Group of Picture unit. In this paper, we then apply the theory to reds nightmare encoded in MPEG, and find minimum smoothing buffer size, initial buffer size. It can be used to study the smoothing of stored video.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.188-198
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• 2000

As MPEG-2 is introduced, a new editing system that works at the compressed domain is required. It is efficient to paste the two different video sequences in the compressed domain, but after repeated pasting, the result sequence can violate MPEG-2 rate control algorithm. In other words, it causes the underflow of the buffer during the decoding time. The earlier study inserted an additional fade-in sequence but it is not natural when we make a program. In this paper, a nobel method using requantization of the underflow sequence is presented.