• Title/Summary/Keyword: Scalable coding

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Exploiting Quality Scalability in Scalable Video Coding (SVC) for Effective Power Management in Video Playback (계층적 비디오 코딩의 품질확장성을 활용한 전력 관리 기법)

  • Jeong, Hyunmi;Song, Minseok
    • KIISE Transactions on Computing Practices
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    • v.20 no.11
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    • pp.604-609
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    • 2014
  • Decoding processes in portable media players have a high computational cost, resulting in high power consumption by the CPU. If decoding computations are reduced, the power consumed by the CPU is also be reduced, but such a choice generally results in a degradation of the video quality for the users, so it is essential to address this tradeoff. We proposed a new CPU power management scheme that can make use of the scalability property available in the H.164/SVC standard. We first proposed a new video quality model that makes use of a video quality metric(VQM) in order to efficiently take into account the different quantization factors in the SVC. We then propose a new dynamic voltage scaling(DVS) scheme that can selectively combine the previous decoding times and frame sizes in order to accurately predict the next decoding time. We then implemented a scheme on a commercial smartphone and performed a user test in order to examine how users react to the VQM difference. Real measurements show that the proposed scheme uses up to 34% fewer energy than the Linux DVFS governor, and user tests confirm that the degradation in the quality is quite tolerable.

An Impact of Addressing Schemes on Routing Scalability

  • Ma, Huaiyuan;Helvik, Bjarne E.;Wittner, Otto J.
    • Journal of Communications and Networks
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    • v.13 no.6
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    • pp.602-611
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    • 2011
  • The inter-domain routing scalability issue is a major challenge facing the Internet. Recent wide deployments of multihoming and traffic engineering urge for solutions to this issue. So far, tunnel-based proposals and compact routing schemes have been suggested. An implicit assumption in the routing community is that structured address labels are crucial for routing scalability. This paper first systematically examines the properties of identifiers and address labels and their functional differences. It develops a simple Internet routing model and shows that a binary relation T defined on the address label set A determines the cardinality of the compact label set L. Furthermore, it is shown that routing schemes based on flat address labels are not scalable. This implies that routing scalability and routing stability are inherently related and must be considered together when a routing scheme is evaluated. Furthermore, a metric is defined to measure the efficiency of the address label coding. Simulations show that given a 3000-autonomous system (AS) topology, the required length of address labels in compact routing schemes is only 9.12 bits while the required length is 10.64 bits for the Internet protocol (IP) upper bound case. Simulations also show that the ${\alpha}$ values of the compact routing and IP routing schemes are 0.80 and 0.95, respectively, for a 3000-AS topology. This indicates that a compact routing scheme with necessary routing stability is desirable. It is also seen that using provider allocated IP addresses in multihomed stub ASs does not significantly reduce the global routing size of an IP routing system.

A Linear Time Algorithm for Constructing a Sharable-Bandwidth Tree in Public-shared Network (공유 네트워크에서 공유대역폭 트리 구성을 위한 선형 시간 알고리즘)

  • Chong, Kyun-Rak
    • Journal of the Korea Society of Computer and Information
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    • v.17 no.6
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    • pp.93-100
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    • 2012
  • In this paper we have proposed a linear time algorithm for solving the minimum sharable-bandwidth tree construction problem. The public-shared network is a user generated infrastructure on which a user can access the Internet and transfer data from any place via access points with sharable bandwidth. Recently, the idea of constructing the SVC video streaming delivery system on public-shared network has been proposed. To send video stream from the stream server to clients on public-shared network, a tree structure is constructed. The problem of constructing a tree structure to serve the video streaming requests by using minimum amount of sharable bandwidth has been shown to be NP-hard. The previously published algorithms for solving this problem are either unable to find solutions frequently or less efficient. The experimental results showed that our algorithm is excellent both in the success rate of finding solutions and in the quality of solutions.

PSNR Comparison of DCT-domain Image Resizing Methods (DCT 영역 영상 크기 조절 방법들에 대한 PSNR 비교)

  • Kim Do nyeon;Choi Yoon sik
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.10C
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    • pp.1484-1489
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    • 2004
  • Given a video frame in terms of its 8${\times}$8 block-DCT coefncients, we wish to obtain a downsized or upsized version of this Dame also in terms of 8${\times}$8 block DCT coefficients. The DCT being a linear unitary transform is distributive over matrix multiplication. This fact has been used for downsampling video frames in the DCT domains in Dugad's, Mukherjee's, and Park's methods. The downsampling and upsampling schemes combined together preserve all the low-frequency DCT coefficients of the original image. This implies tremendous savings for coding the difference between the original frame (unsampled image) and its prediction (the upsampled image).This is desirable for many applications based on scalable encoding of video. In this paper, we extend the earlier works to various DCT sizes, when we downsample and then upsample of an image by a factor of two. Through experiment, we could improve the PSM values whenever we increase the DCT block size. However, because the complexity will be also increase, we can say there is a tradeoff. The experiment result would provide important data for developing fast algorithms of compressed-domain image/video resizing.

A study of Development of Transmission Systems for Terrestrial Single Channel Fixed 4K UHD & Mobile HD Convergence Broadcasting by Employing FEF (Future Extension Frame) Multiplexing Technique (FEF (Future Extension Frame) 다중화 기법을 이용한 지상파 단일 채널 고정 4K UHD & 이동 HD 융합방송 전송시스템 개발에 관한 연구)

  • Oh, JongGyu;Won, YongJu;Lee, JinSeop;Kim, JoonTae
    • Journal of Broadcast Engineering
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    • v.20 no.2
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    • pp.310-339
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    • 2015
  • In this paper, the possibility of a terrestrial fixed 4K UHD (Ultra High Definition) and mobile HD (High Definition) convergence broadcasting service through a single channel employing the FEF (Future Extension Frame) multiplexing technique in DVB (Digital Video Broadcasting)-T2 (Second Generation Terrestrial) systems is examined. The performance of such a service is also investigated. FEF multiplexing technology can be used to adjust the FFT (fast Fourier transform) and CP (cyclic prefix) size for each layer, whereas M-PLP (Multiple-Physical Layer Pipe) multiplexing technology in DVB-T2 systems cannot. The convergence broadcasting service scenario, which can provide fixed 4K UHD and mobile HD broadcasting through a single terrestrial channel, is described, and transmission requirements of the SHVC (Scalable High Efficiency Video Coding) technique are predicted. A convergence broadcasting transmission system structure is described by employing FEF and transmission technologies in DVB-T2 systems. Optimized transmission parameters are drawn to transmit 4K UHD and HD convergence broadcasting by employing a convergence broadcasting transmission structure, and the reception performance of the optimized transmission parameters under AWGN (additive white Gaussian noise), static Brazil-D, and time-varying TU (Typical Urban)-6 channels is examined using computer simulations to find the TOV (threshold of visibility). From the results, for the 6 and 8 MHz bandwidths, reliable reception of both fixed 4K UHD and mobile HD layer data can be achieved under a static fixed and very fast fading multipath channel.

Implementation of Internet Terminal using G.729.1 Wideband Speech Codec for Next Generation Network (차세대 통신망을 위한 G.729.1 광대역 음성 코덱을 활용한 인터넷 단말 구현)

  • So, Woon-Seob;Kim, Dae-Young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.10B
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    • pp.939-945
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    • 2008
  • Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.