• Title/Summary/Keyword: video delay

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Latency Analysis of AVB Network and Optimization Design for Automotive

  • An, Byoungman;Kim, YoungSeop
    • Journal of the Semiconductor & Display Technology
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    • v.18 no.3
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    • pp.127-132
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    • 2019
  • This paper presents an overview of automotive communication technologies, including related technology developments. We describe the latency of Audio Video Bridge (AVB) network as well as purpose the optimized design of the Ethernet network system for automotive. Our design plays a significant role in reducing the delay between components. The proposed approach on realistic test cases showed that there was a delay reduction, approximately 49.4%. It is expected that the optimization method for the actual automotive environment can greatly shorten the time period in the design and development process. The results obtained from the experiments on the delay time present in each function are reliable because average values are obtained through repeated actual tests for several months. It will greatly benefit the industry since analyzing the latency between each function in a short period of time is very important.

An Improvement in Intra-Slice Low Delay Video Coding for Digital TV Broadcasting (디지틀 TV 방송을 위한 저지연 intra-slice 영상 부호화 방식의 개선 방법)

  • 권순각;김재균
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.19 no.12
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    • pp.2376-2385
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    • 1994
  • In receiving the digital TV signal, both decoding delay and the channel hopping delay are very critical factors in applications. The intra-slice coding in the MPEG-2 SIMPLE PROFLE of No B-picture is one of the primary methods for short delay time in video decoding. It has the advantage of short decoding delay, but has the drawback of long channel hopping delay time. In this paper, we propose a method to reduce the channel delay with negligible loss in SNR performance. It is shown that dividing pictures into several regions of slices and adding some restriction in motion vector search for inter-frame coding. hence the random acess points are effectively increased. and the channel hopping delay is reduced.

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SPMLD: Sub-Packet based Multipath Load Distribution for Real-Time Multimedia Traffic

  • Wu, Jiyan;Yang, Jingqi;Shang, Yanlei;Cheng, Bo;Chen, Junliang
    • 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.

Design and Implementation of a Mobile Terminal Device for Video Transmission in Multiple Networks (다중 네트워크에서의 동영상 전송을 위한 이동 단말 장치의 설계 및 구현)

  • Kim, Jong-Heon;Lee, Dong-Soo
    • The Journal of The Korea Institute of Intelligent Transport Systems
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    • v.12 no.2
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    • pp.38-44
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    • 2013
  • In this paper, we designed and implemented a mobile video transmission device with 1080i resolution and 30fps frame rate using wireless multi-access technologies. The video acquisition delay time of 16us and jitter of 21ms for this equipment were obtained, respectively. The device is designed to access various communication networks. In order to evaluate the performance of its video transmission capability, a server and monitoring viewer are implemented. In the performance test using LTE network, 211ms delay time was obtained at 3Mbps transmission. Rendering of 1fps frame rate at the 265kbps transmission was confirmed for the transmission test using Inmarsat network.

Performance Comparison of HEVC and H.264/AVC Standards in Broadcasting Environments

  • Dissanayake, Maheshi B.;Abeyrathna, Dilanga L.B.
    • Journal of Information Processing Systems
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    • v.11 no.3
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    • pp.483-494
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    • 2015
  • High Efficiency Video Coding (HEVC) is the most recent video codec standard of the ITU-T Video Coding Experts Group and the ISO/IEC Moving Picture Experts Group. The main goal of this newly introduced standard is for catering to high-resolution video in low bandwidth environments with a higher compression ratio. This paper provides a performance comparison between HEVC and H.264/AVC video compression standards in terms of objective quality, delay, and complexity in the broadcasting environment. The experimental investigation was carried out using six test sequences in the random access configuration of the HEVC test model (HM), the HEVC reference software. This was also carried out in similar configuration settings of the Joint Scalable Video Module (JSVM), the official scalable H.264/AVC reference implementation, running on a single layer mode. According to the results obtained, the HM achieves more than double the compression ratio compared to that of JSVM and delivers the same video quality at half the bitrate. Yet, the HM encodes two times slower (at most) than JSVM. Hence, it can be concluded that the application scenarios of HM and JSVM should be judiciously selected considering the availability of system resources. For instance, HM is not suitable for low delay applications, but it can be used effectively in low bandwidth environments.

A Study on the Traffic Delay Control for real-time Communication in Broadband Network (광대역망에서 실시간 통신을 위한 트래픽 지연제어에 관한 연구)

  • Jang, Hyok;Kim, Kwang-Jun;Na, Sang-Dong;Cho, Hyeon-Seob;Jang, Sung-Whan;Bae, Chul-Soo
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06c
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    • pp.351.1-355
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    • 1998
  • Network-based real-time multimedia applications require guaranteed performances of communication service. Also according to increased bandwidth and processing performance in the future high speed networks, it is possible to make multimedia real-time applications such as distributed real-time applications including video conference, medical imaging, and video education. These applications have characteristics of stringent performance requirements in terms of delay, delay jitter and maximum loss rate. This paper proposes a new delay control for Broadband networks with FCFS queueing. The delay control provides an alternative solution to a current problem in real-time communication; traffic distortion, which makes it difficult to meet such real-time requirements. In simulations, it shows that the proposed delay control can guarantee local and end-to-end delay.

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Removal of Complexity Management in H.263 Codec for A/VDelivery Systems

  • Jalal, Ahmad;Kim, Sang-Wook
    • 한국HCI학회:학술대회논문집
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    • 2006.02a
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    • pp.931-936
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    • 2006
  • This paper presents different issues of the real-time compression algorithms without compromising the video quality in the distributed environment. The theme of this research is to manage the critical processing stages (speed, information lost, redundancy, distortion) having better encoded ratio, without the fluctuation of quantization scale by using IP configuration. In this paper, different techniques such as distortion measure with searching method cover the block phenomenon with motion estimation process while passing technique and floating measurement is configured by discrete cosine transform (DCT) to reduce computational complexity which is implemented in this video codec. While delay of bits in encoded buffer side especially in real-time state is being controlled to produce the video with high quality and maintenance a low buffering delay. Our results show the performance accuracy gain with better achievement in all the above processes in an encouraging mode.

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A Study on the Performance Enhancements of Video Streaming Service in MPLS Network

  • Kwak Kyoung Hwan;Park In Kap;Kim Chung Hyun
    • Proceedings of the IEEK Conference
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    • 2004.08c
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    • pp.549-551
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    • 2004
  • This paper used OPNET to simulate video streaming service a test IP network and MPLS network with the traffic shaping that have with CQ_ LLQ algorithm, LSP of fixed bandwidth, policy of limitation users and measures parameters such as delay, throughput, packet loss. To verify the performance of video streaming service in IP network and MPLS network, two scenario that have same topology and traffic source. One is the simulation for best-effort service in pure IP network. The other is the simulation for QoS-enabled service in MPLS Network. Based on simulation result, the MPLS network with CQ_ LLQ algorithm and fixed LSP show advantage of the video streaming service QoS, specially delay and packet loss

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Stereo-video Synchronization for 3D Video Transmission (3차원 비디오 전송을 위한 스테레오비디오 동기화 방법)

  • Lee, Dong-Jin;Lee, Seon-Oh;Sim, Dong-Gyu;Lee, Hyuk-Joon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.4B
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    • pp.349-359
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    • 2009
  • In this paper, we propose a stereo-video transmission method for reduction of delay and maximization of 3D effect. Conventional multimedia synchronization algorithms were designed to achieve minimum delay and synchronize multiple video and audio streams, however, they could not be effective for 3D video transmission. In this paper, we proposed a synchronization algorithm by considering the minimum error of time difference between streams for 3D effect. The minimum error of time difference for 3D effect was derived based on a 3D subjective quality test. We compute display time of the delivered videos within the allowed time-difference and the video are displayed according to the display time. To evaluate the performance of the proposed algorithm, we implemented a real-time video communication system and subjective quality test has been conducted with the proposed system. We found that video quality displayed by the proposed system. We found that video quality displayed by the proposed algorithm ranks 'good' and 'excellent' in the DMOS (Differential Mean Opinion Score) scale, based on the MOS (Mean Opinion Score) test.

Distortion Variation Minimization in low-bit-rate Video Communication

  • Park, Sang-Hyun
    • Journal of information and communication convergence engineering
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    • v.5 no.1
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    • pp.54-58
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    • 2007
  • A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for distortion variation minimization. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. The proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performances than the existing rate control method.