• ETRI Journal
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• v.43 no.3
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• pp.447-458
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• 2021

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3856-3872
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• 2018

HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.721-732
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• 2000

Multimedia communication systems require not only high-performance computer hardwares and high-speed networks, but also a buffer management mechanism to process many data efficiently. Two buffer handling methods, Push and Pull, are commonly used. In the Push method, a server controls the flow of dat to a client, while in the Pull method, a client controls the flow of data from a server. Those buffering schemes can be applied to the data transfer between the packet receiving buffer, which receives media data from a network server, and media playout devices, which play the recived media data. However, the buffer management mechanism in client-sides mainly support either one of the Push or the Pull method. Consequently, they have some limitations to support various media playout devices. Futhermore, even though some of them support both methods, it is difficult to use since they can't provide a unified structure. To resolved these problems, in this paper, we propose an efficient and flexible Push/Pull buffer management mechanism at client-side. The proposed buffer management scheme supports both Push and Pull method to provide various media playout devices and to support buffering function to absorb network jitter. The proposed scheme can support the various media playback devices using a single buffer space which in consequence, saves memory space compared to the case that a client keeps tow types of buffers. Moreover, it facilitates the single buffer as a mechanism for the absorbing network jitter effectively and efficiently. The proposed scheme has been implemented in an existing multimedia communication system, so called ISSA (Integrated Streaming Service Architecture), and it shows a good performance result compared to the conventional buffering methods in multimedia communication environments.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.413-417
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• 2009

The use of scalable video coding scheme has been regarded as a promising solution for guaranteeing the quality of service of the video streaming over the Internet because it is a capable coding scheme to perform quality adaptation depending on network conditions. In this paper, we use a streaming model that transmits base layer using TCP and enhancement layers using DCCP, which try to provide transmission reliability of the BL and TCP friendliness. Unlike pervious works, the proposed algorithm performs rate adaptation based on playout buffer status. The PoB status of the client is sent back periodically to the server and serves as a network congestion indicator. Experimental results show that our scheme improves streaming quality comparing with pervious scheme in the case of not only constant/dynamic background flows but also VBR-encoded video sequence.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.3
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• pp.48-54
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• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.4
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• pp.57-67
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• 1996

This paper aims to support delay-sensitive multimedia applications by suggesting a mechanism in which maintains almost constant end-to-end delay thus providing the optimum playout synchronization. For this task the sum of network delay and buffering delay is entiredly managed and to eliminate little delay fluctuations and instantaneous delays at a buffer and a network, a low pass filter is used. Furthermore the correction function, which is used for maintaining the buffering level ot a reference value, is a non-linear step function, unlike the existing linear and continuous function. it has a different step sizes adapting to a traffic characteristics of a network congestion. the proposed mechanism has been confirmed of it sefficiency through SLAM-II netowrk.

• 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.