• Title/Summary/Keyword: Playback buffer

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A Dynamic Bandwidth Allocation Scheme based on Playback Buffer Level in a Distributed Mobile Multimedia System (분산 모바일 멀티미디어 시스템에서 재생 버퍼 수준에 기반한 동적 대역폭 할당 기법)

  • Kim, Jin-Hwan
    • The KIPS Transactions:PartB
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    • v.17B no.6
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    • pp.413-420
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    • 2010
  • In this paper, we propose a scheme for dynamic allocating network bandwidth based on the playback buffer levels of the clients in a distributed mobile multimedia system. In this scheme, the amount of bandwidth allocated to serve a video request depends on the buffer level of the requesting client. If the buffer level of a client is low or high temporarily, more or less bandwidth will be allocated to serve it with an objective to make it more adaptive to the playback situation of this client. By employing the playback buffer level based bandwidth allocation policy, fair services can also be provided to the clients. In order to support high quality video playbacks, video frames must be transported to the client prior to their playback times. The main objectives in this bandwidth allocation scheme are to enhance the quality of service and performance of individual video playback such as to minimize the number of dropped video frames and at the same time to provide fair services to all the concurrent video requests. The performance of the proposed scheme is compared with that of other static bandwidth allocation scheme through extensive simulation experiments, resulting in the 4-9% lower ratio of frames dropped according to the buffer level.

Impact of playout buffer dynamics on the QoE of wireless adaptive HTTP progressive video

  • Xie, Guannan;Chen, Huifang;Yu, Fange;Xie, Lei
    • 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.

A Novel Bit Rate Adaptation using Buffer Size Optimization for Video Streaming

  • Kang, Young-myoung
    • International Journal of Internet, Broadcasting and Communication
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    • v.12 no.4
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    • pp.166-172
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    • 2020
  • Video streaming application such as YouTube is one of the most popular mobile applications. To adjust the quality of video for available network bandwidth, a streaming server provides multiple representations of video of which bit rate has different bandwidth requirements. A streaming client utilizes an adaptive bit rate scheme to select a proper video representation that the network can support. The download behavior of video streaming client player is governed by several parameters such as maximum buffer size. Especially, the size of the maximum playback buffer in the client player can greatly affect the user experience. To tackle this problem, in this paper, we propose the maximum buffer size optimization according to available network bandwidth and buffer status. Our simulation study shows that our proposed buffer size optimization scheme successfully mitigates playback stalls while preserving the similar quality of streaming video compared to existing ABR schemes.

Estimating the Optimal Buffer Size on Mobile Devices for Increasing the Quality of Video Streaming Services (동영상 재생 품질 향상을 위한 최적 버퍼 수준 결정)

  • Park, Hyun Min
    • The Journal of the Korea Contents Association
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    • v.18 no.3
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    • pp.34-40
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    • 2018
  • In this study, the optimal buffer size is calculated for seamless video playback on a mobile device. Buffer means the memory space for multimedia packet which arrives in mobile device for video play such as VOD service. If the buffer size is too large, latency time before video playback can be longer. However, if it is too short, playback service can be paused because of shortage of packets arrived. Hence, the optimal buffer size insures QoS of video playback on mobile devices. We model the process of buffering into a discret-time queueing model. Mean busy period length and mean waiting time of Geo/G/1 queue with N-policy is analyzed. After then, we uses the main performance measures to present numerical examples to decide the optimal buffer size on mobile devices. Our results enhance the user satisfaction by insuring the seamless playback and minimizing the initial delay time in VOD streaming process.

Adaptive Load Balancing Scheme for Real-Time Video Stream Transmission in Mobile Environment (모바일 환경에서 실시간 비디오 스트림 전송을 위한 적응형 부하 조정 기법)

  • Kim, Jin-Hwan
    • 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.

ZigBee-based Real-time Wireless Networked Motor Control System (지그비 기반의 실시간 무선 네트워크 모터 제어시스템)

  • Park, Jung-Il
    • IEMEK Journal of Embedded Systems and Applications
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    • v.15 no.2
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    • pp.103-109
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    • 2020
  • This paper finds solutions for using ZigBee in wireless networked control system (WNCS). The round trip time delay and packet loss rate of the WNCS are measured. On the basis of these measured data, a playback buffer is used to solve the variable time delay in WNCS, and a Smith predictor is introduced to compensate for the time delay. The WNCS was able to be actually constructed to perform DC motor position control with 40 Hz sampling frequency.

A Frame Skipping Transfer Policy for Server based on Clients Dynamic Buffer Control Signal (클라이언트의 동적 버퍼 제어 신호에 의한 서버의 프레임 생략 전송 정책)

  • Jeong, Hong-Seup;Park, Kyoo-Seok
    • The KIPS Transactions:PartA
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    • v.10A no.6
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    • pp.677-684
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    • 2003
  • Most policies of VOD services are mainly studied about server system, and make no touch on clients buffer stabilization which is the basis of guaranteeing playback quality and playing without cut off. In this paper, we proposed a skipping frame transfer policy that can sending dynamic buffer control signal to server in accordance with clients buffer state of starvation or overrun, and server transfers a skipped frame to client after receive the control signal. And through a simulation, we show the suggested policy is more efficiency on playback quality by buffer stabilization which is by preventing loss and miss for receiving frame due to grow worse client buffer state.

Initial Buffering-Time Decision Scheme for Progressive Multimedia Streaming Service (프로그레시브 멀티미디어 스트리밍 서비스를 위한 초기 버퍼링 시간 결정 기법)

  • Seo, Kwang-Deok;Jung, Soon-Heung
    • Journal of KIISE:Computing Practices and Letters
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    • v.14 no.2
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    • pp.206-210
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    • 2008
  • The most noticeable aspect of progressive streaming is the media playback during its download through TCP to avoid a lengthy wait for a content to finish downloading. By employing TCP, it is usually possible to detect lost packets by using the checksum and sequence numbering functions of TCP Thereafter, we can recover the lost packets by the retransmission function of TCP. However, there must remain enough amount of media data in the recipient buffer in order to guarantee seamless media playback even during retransmission. In this paper, we propose an efficient algorithm for determining the initial buffering time before start of playback to guarantee seamless playback during retransmission considering the probability of client buffer under-flow. The effectiveness of the proposed algorithm will be proved through extensive simulation results.

Adaptive Service Mode Conversion to Minimize Buffer Space Requirement in VOD Server (주문형 비디오 서버의 버퍼 최소화를 위한 가변적 서비스 모드 변환)

  • Won, Yu-Jip
    • Journal of KIISE:Computer Systems and Theory
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    • v.28 no.5
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    • pp.213-217
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    • 2001
  • Excessive memory buffer requirement in continuous media playback is a serious impediment of wide spread usage of on-line multimedia service. Skewed access frequency of available video files provides an opportunity of re-using the date blocks which has been loaded by one session for later usage. We present novel algorithm which minimizes the buffer requirement in multiple sessions of multimedia playbacks. In continuous media playback originated from the disk, a certain amount of memory buffer is required to synchronize asynchronous disk. Read operation and synchronous playback operation. As aggregate playback bandwodth increases, larger amount of buffer needs to be allocated for this synchronization purpose. The focus of this work is to study the asymptotic behavior of the synchronization buffer requirement and to develop an algorithm coping with this excessive buffer requirement under bandwidth congestioon. We argue that in a large scale continuous media server, it may not be necessary to read the blocks for each session directly from the disk. The beauty of our work lies in the fact that it dynamically adapts to disk utilization of the server and finds the optimal way of servicinh the individual sessions while minimizing the overall buffer space requirement. Optimality of the proposed algorithm is shown by proof. The effectiveness and performance of the proposed scheme is examined via simulation.

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Mobile Client Buffer Level-based Scheduling Algorithms for Variable-Bit-Rate Video Stream Transmission (VBR 비디오 스트림 전송을 위한 모바일 클라이언트 버퍼 수준 기반 스케쥴링 알고리즘)

  • Kim, Jin-Hwan
    • Journal of Korea Multimedia Society
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    • v.15 no.6
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    • pp.814-826
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    • 2012
  • In this paper, we propose scheduling algorithms for transporting variable-bit-rate video stream using playback buffer level of the clients over wireless communication networks. The proposed algorithms attempt to maximize the utilization of limited bandwidth between the central video server and the clients over a mobile network. Since a video server may serve several video request at the same time, it is important to allocate and utilize network bandwidth to serve them fairly and efficiently. In order to improve the quality of service and real-time performance of individual video playback, the video server attempts to allocate temporarily more network bandwidth to serve a video request with the lower buffer level preferentially. The simulation results prove the fair service and load balancing among the mobile concurrent clients with different buffer levels and hence maximizing the number of frames that are transported successfully to the client prior to their playback times.