• 제목/요약/키워드: Bandwidth Adaptation

검색결과 73건 처리시간 0.027초

Adaptive Transcoding for Object-based MPEG-4 Scene using Optimal Configuration of Objects

  • Cha, Kyung-Ae
    • 한국멀티미디어학회논문지
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    • 제9권12호
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    • pp.1560-1571
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    • 2006
  • In order to transmit multimedia streams over the network with a timely changing channel bandwidth such as Internet, scalable video coding schemes have been studied to represent video in flexible bitstream. Much research has been made on how to represent encoded media(such as video) bitstream in scalable ways. In this paper, rte propose an optimal selection of the objects for MPEG-4 bitstream adaptation to meet a given constraint. We adopt a multiple choice knapsack problem with multi-step selection for the MPEG-4 objects with different bit-rate scaling levels in the MPEG-4 bitstream. The bitstream adaptation based on the optimal selection result is then to fetch the necessary parts of the MPEG-4 bitstream to constitute an adapted version of the original MPEG-4 binary resource. The experiment results show that the optimal selection of MPEG-4 objects for a given constraint can promisingly be made which meets the given constraint.

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Optimizing Fsync Performance with Dynamic Queue Depth Adaptation

  • Park, Daejun;Kim, Min Ji;Shin, Dongkun
    • JSTS:Journal of Semiconductor Technology and Science
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    • 제15권5호
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    • pp.570-576
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    • 2015
  • Existing flash storage devices such as universal flash storage and solid state disk support command queuing to improve storage I/O bandwidth. Command queuing allows multiple read/write requests to be pending in a device queue. Because multi-channel and multi-way architecture of flash storage devices can handle multiple requests simultaneously, command queuing is an indispensable technique for utilizing parallel architecture. However, command queuing can be harmful to the latency of fsync system call, which is critical to application responsiveness. We propose a dynamic queue depth adaptation technique, which reduces the queue depth if user application is expected to send fsync calls. Experiments show that the proposed technique reduces the fsync latency by 79% on average compared to the original scheme.

원격 의료 진단 시스템을 위한 사용자 기반 적응 대역폭 비디오 시스템 (A User Driven Adaptable Bandwidth Video System for Remote Medical Diagnosis System)

  • 정영지;더스틴 라이트;유수프 어즈터크
    • 한국IT서비스학회지
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    • 제14권1호
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    • pp.99-113
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    • 2015
  • Adaptive bitrate (ABR) streaming technology has become an important and prevalent feature in many multimedia delivery systems, with content providers such as Netflix and Amazon using ABR streaming to increase bandwidth efficiency and provide the maximum user experience when channel conditions are not ideal. Where such systems could see improvement is in the delivery of live video with a closed loop cognitive control of video encoding. In this paper, we present streaming camera system which provides spatially and temporally adaptive video streams, learning the user's preferences in order to make intelligent scaling decisions. The system employs a hardware based H.264/AVC encoder for video compression. The encoding parameters can be configured by the user or by the cognitive system on behalf of the user when the bandwidth changes. A cognitive video client developed in this study learns the user's preferences (i.e. video size over frame rate) over time and intelligently adapts encoding parameters when the channel conditions change. It has been demonstrated that the cognitive decision system developed has the ability to control video bandwidth by altering the spatial and temporal resolution, as well as the ability to make scaling decisions

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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    • 제12권4호
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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.

균일 및 주파수 선택적 페이딩에서 대역폭 효율의 적응 QAM 성능분석 (Performance Evaluation of Bandwidth Efficient Adaptive QAM Schemes in Flat and Frquency Selective Fading Channels)

  • 정연호
    • 한국통신학회논문지
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    • 제25권10A호
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    • pp.1473-1479
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    • 2000
  • This paper presents the performance evaluation of an adaptive QAM scheme under flat and frequency selective fading channels for indoor wireless communication systems. The QAM modulation is combined with differential encoding and the demodulation process is carried out noncoherently. The adaptation is performed by varying the modulation level of QAM, depending upon received signal strength. The adaptation mechanism allows a 2- or 3-bit increase or decrease at a time, if the channel condition is considered to be significantly good or bad. Simulation results show that the average number of bits per symbol (ABPS) for each symbol block transmitted over a flat fading channel is higher than 5.0 and the BER performance is better than 10^-4 for a SNR value higher than 30 dB. For frequency selective fading channels, an oversampling technique in the receiver was employed. The BER performance obtained for frequency selective fading channels is better than 10^-4 with a SNR value of 40 dB and ABPS is found to be approximately 5.5. Therefore, this scheme is very useful in that it provides both very high bandwidth efficiency and acceptable performance with moderate SNR values over flat and frequency selective fading channels. In addition, this scheme provides reduced receiver complexity by way of noncoherent detection.

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네트워크 혼잡상태에 동적 적응을 위한 계층적 멀티캐스트 (A Hierarchical Multicast for Dynamic Adaptation to Network Congestion Status)

  • 김창근;송진국;구명모
    • 한국정보통신학회논문지
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    • 제12권8호
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    • pp.1427-1433
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    • 2008
  • 멀티캐스트 응용에서 네트워크 혼잡상태에 동적 적응을 위한 방법으로 SARLM이 있다. 이 방법에서는 혼잡이 발생하면 혼잡지역의 전송률 감소로 인하여 비 혼잡지역에서 가용 대역폭의 낭비가 발생한다. 본 논문에서는, 네트워크 혼잡에 동적 적응을 위한 계층적 멀티캐스트를 제안한다. 제안한 방법에서는 지역마다 대표자를 선택하여 혼잡상태일 때, 전송률 감소를 방지하기 위하여 멀티캐스트 송신자로부터 전송된 패킷을 혼잡지역 대표자에게 유니캐스트로 계층적 전송하고, 혼잡지역 대표자는 지역 수신자들에게 멀티캐스트로 전송한다. 실험 결과, 제안기법이 혼잡상태에서 수신자들의 전송률을 향상시키고 가용 대역폭을 보다 더 효율적으로 이용함을 알 수 있었다.

적응형 서비스를 제공하는 이동통신망에서의 서비스 품질 척도 추정 모델 (Quality of Service Parameters Estimation Model for Adaptive Bandwidth Service in Mobile Cellular Networks)

  • 정성환;홍정완;이창훈
    • 대한산업공학회지
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    • 제33권1호
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    • pp.153-163
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    • 2007
  • An adaptive framework paradigm where the bandwidth values of the ongoing calls vary according to the trafficsituations is one of the promising concepts for overcoming poor resource conditions due to handoffs in mobilecellular networks. However, quantifying the level of bandwidth degradation of the ongoing calls in an adaptiveframework is important in view of Quality of Service (QoS) Provisioning. Therefore we introduce new QoSparameters, the Degradation Degree Ratio (DDR), which represents the average portion of the degradationdegree during degradation pehod of a call, and the Degradation Area Ratio (DAR), which represents the averageratio of a call's degradation level considering both the period and the degree of degradation jointly in multi-levelbandwidth service. We also develop a new analytical model for estimating the QoS measures such as theDegradation Pehod Ratio (DPR), DDR and DAR. We show how to calculate the QoS measures and illustrate themethod by numerical examples. The proposed model can be used to determine the optimal parameter of theCAC scheme and analyze the sensitivity ofthe QoS parameters in adaptive networks.

Switching Picture Added Scalable Video Coding and its Application for Video Streaming Adaptive to Dynamic Network Bandwidth

  • Jia, Jie;Choi, Hae-Chul;Kim, Hae-Kwang
    • 방송공학회논문지
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    • 제13권1호
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    • pp.119-127
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    • 2008
  • Transmission of video over Internet or wireless network requires coded stream capable of adapting to dynamic network conditions instantly. To meet this requirement, various scalable video coding schemes have been developed, among which the Scalable Video Coding (SVC) extension of the H.264/AVC is the most recent one. In comparison with the scalable profiles of previous video coding standards, the SVC achieves significant improvement on coding efficiency performance. For adapting to dynamic network bandwidth, the SVC employs inter-layer switching between different temporal, spatial or/and fidelity layers, which is currently supported with instantaneous decoding refresh (IDR) access unit. However, for real-time adaptability, the SVC has to frequently employ the IDR picture, which dramatically decreases the coding efficiency. Therefore, an extension of SP picture from the AVC to the SVC for an efficient inter-layer switching is investigated and presented in this paper. Simulations regarding the adaptability to dynamic network bandwidth are implemented. Results of experiment show that the SP picture added SVC provides an average 1.2 dB PSNR enhancement over the current SVC while providing similar adaptive functionality.

Loss-RTT 기반 차등 전송률 조절 알고리즘에 관한 연구 (Loss-RTT based Differentiated Rate Adaptation Algorithm)

  • 김지언;정재일
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2000년도 추계종합학술대회 논문집(1)
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    • pp.17-20
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    • 2000
  • TCP is ill-suited to real-time multimedia applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. Deploying non congestion controlled traffic results in extreme unfairness towards competing TCP traffic. Therefore, they need to be enhanced with congestion control schemes that not only am at reducing loss ratios and improve bandwidth utilization but also are fair towards competing TCP connections. This paper proposes a differentiated rate adaptation algorithm based on loss and round trip time. Rate in a sender quickly responds to loss ratio and holds steady state. Additionally, this algorithm reduces loss ratio by loss prediction in a receiver.

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Rate Adaptation with Q-Learning in CSMA/CA Wireless Networks

  • Cho, Soohyun
    • Journal of Information Processing Systems
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    • 제16권5호
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    • pp.1048-1063
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    • 2020
  • In this study, we propose a reinforcement learning agent to control the data transmission rates of nodes in carrier sensing multiple access with collision avoidance (CSMA/CA)-based wireless networks. We design a reinforcement learning (RL) agent, based on Q-learning. The agent learns the environment using the timeout events of packets, which are locally available in data sending nodes. The agent selects actions to control the data transmission rates of nodes that adjust the modulation and coding scheme (MCS) levels of the data packets to utilize the available bandwidth in dynamically changing channel conditions effectively. We use the ns3-gym framework to simulate RL and investigate the effects of the parameters of Q-learning on the performance of the RL agent. The simulation results indicate that the proposed RL agent adequately adjusts the MCS levels according to the changes in the network, and achieves a high throughput comparable to those of the existing data transmission rate adaptation schemes such as Minstrel.