適應性編碼技術基於服務品質之丟棄封包機制於IEEE 802.16e網路之研究

Video communication requires sufficient resources to obtain the desired image quality. Availability of bandwidth and screen resolution from different devices on the client end, as well as compression techniques used for transmission also significantly influence to the video quality. Even though the H.264/AVC video compression standard and its scalable extension is known as Scalable Video Coding (SVC) was introduced and successfully addressed the challenge, some weaknesses persist especially for high-resolution video transmissions. The size of the bitstream generated by the SVC standard for ultra-high resolution video compression can only be met if transmitted on a network with a large bandwidth. The Scalable High-Efficiency Video Coding (SHVC) were introduced to overcome problems in the SVC standards. The new video compression standard is proven to reduce bandwidth requirements by up to 46.41 % compared to the SVC standards for the same video quality. This study introduces the performance analysis of SHVC standard video transmission as compared to SVC standard. Both compression standards are transmitted and emulated through network emulator called Mininet-Wifi to get the real transmission picture. The client-server video transmission scheme with different bandwidth and various client position are set in the experiment scenario. It is found that transmission time, packet loss, and video reconstruction quality on scalable video transmission with SHVC standard is better than scalable video transmissions with SVC standard.

In heterogeneous network environments, it is mandatory to measure the grade of the video quality in order to guarantee the optimal quality of the video streaming service. Quality of Service (QoS) has become a key issue for service acceptability and user satisfaction. Although there have been many recent works regarding video quality, most of them have been limited to measuring quality within temporal and Signal-to-Noise Ratio (SNR) scalability. H.264/AVC Scalable Video Coding (SVC) has emerged and has been developed to support full scalability. This includes spatial, temporal, and SNR scalability, each of which shows different visual effects. The aim of this paper is to define and develop a novel video quality metric allowing full scalability. It focuses on the effect of frame rate, SNR, the change of spatial resolution, and motion characteristics using subjective quality assessment. Experimental results show the proposed quality metric has a high correlation to subjective quality and that it is useful in determining the video quality of SVC.

In this paper, we address the problem of QoS (Quality of Service) provisioning for multi-user video streaming over multiple heterogeneous wireless networks based on the distributed, cross-layer design framework. By jointly considering the rate allocation and the Joint Source-Channel Coding (JSCC), our proposal aims at maximizing the QoS provisioning under the given resource constraint. At first, we develop and evaluate a framework for optimal video rate allocation over multiple networks based on the observed Available Bit Rate (ABR) and the Round Trip Time (RTT) over each access network, as well as the video rate-distortion characteristics. The rate allocation is formulated as a convex optimization problem that minimizes the sum of all video streams expected distortion. Then, we propose an analytical JSCC scheme for error-resilient scalable encoded video, and integrate the JSCC with the specific rate allocation algorithm to improve the constructed video quality by optimally applying the appropriate channel coding rate given the constraints imposed by the transmission rate and the prevailing channel conditions. Objective and subjective simulation results are provided which demonstrate the effectiveness of our proposed joint scheme.

Scenarios for the use of the recently approved Scalable Video Coding (SVC) extension of H.264/MPEG4-AVC in IPTV services are presented. For that, a brief technical overview of SVC when deployed in IPTV services is provided. The coding efficiency of the various scalability types of SVC is demonstrated followed by an analysis of the complexity of the various SVC tools. Based on this technical characterization, it is described how the different SVC features such as efficient methods for graceful degradation, bit rate adaptation, and format adaptation, can be mapped to application requirements of IPTV services. It is discussed how such mappings can lead to improved content portability, management and distribution as well as an improved management of access network throughput resulting in better quality of service and experience for the users of IPTV services.

Software-Defined Networking (SDN) is a new efficiently idea of programmable networks that separates the control plane from data plane of all network devices. Internet service provider is responsible for all the control decisions and communication among the forwarding elements from centralized controller. SDN provides the various optimized services. Quality of service (QoS) routing is a path computation method that is suitable for the different traffics generated by several applications, while utilization of network resources has increased. This agreement of service is defined by QoS requirements such as throughput, delay, jitter and packet loss etc. Multimedia applications often require assured from multi QoS constrained, causing the NP-complete problem which cannot be simply solved in polynomial time and high management complexity in the transition network. SDN is able to reduce complexity and it is used to efficiently implement traffic, hence SDN significantly values to development QoS routing. In this paper, we propose QoS routing algorithm called Quantized Level Balance (QLB) for SDN that considers one or many QoS parameters relating to the network application. To satisfy the requirements, QLB selects QoS parameters depending to the level of appropriate application service quality. We have replicated our algorithm on simulate topology with Scalable Video-streaming Evaluation Framework (SVEF). We measure the Peak Signal-to-Noise Ratio (PSNR) and Mean Opinion Score (MOS) of Scalable Video Coding (SVC) at the receiver. Our propose algorithm is improved than single-metric approach that may choose poor QoS parameter paths.

The Scalable Video Coding (SVC) compression technique is extension of H.264/AVC which is developed by Joint Video Team of the ITU-T VCEG and the ISO/IEC MPEG. SVC enables scalability support and capable to adapt bitrate with lower temporal or spatial resolutions for wireless channel bandwidth those varying with time. In this paper, we proposed a cross-layer mechanism, which includes video packet prioritization and Quality of Services (QoS) mapping, for the transmitting SVC video over the IEEE 802.11n wireless networks without aggregation. The proposed mechanism delivery enables interaction with Medium Access Control (MAC) layer, providing controlling of different Access Classifications (ACs) for video packets. Our purposed Traffic Queue Controlling Cross-layer Mechanism (TQCCLM) performs with the SVC packet prioritization at the MAC layer according to Temporal Identification (TId) Quality Identification (QId) values of incoming video SliceData and access queue length. Simulation results shows that the proposed mechanism improves SVC video quality when compared to IEEE 802.11n EDCA default mapping method.

The transmission of video data is a major part of traffic on today's Internet. Since the Internet is a highly dynamic environment, quality adaptation is essential in matching user device resources with the streamed video quality. This can be achieved by applying mechanisms that follow the Scalable Video Coding (SVC) standard, which enables scalability of the video quality in multiple dimensions. In SVC-based streaming, adaptation decisions have long been driven by Quality of Service (QoS) metrics, such as throughput. However, these metrics do not well match the way human users perceive video quality. Therefore, in this paper, the classical SVC-based video streaming approach is expanded to consider Quality of Experience (QoE) for adaptation decisions. The video quality is assessed using existing objective techniques with a high correlation to the human perception. The approach is evaluated in context of a P2P-based Video-on-Demand (VoD) system and shows that by making peers favor always layers with a high estimated QoE but not necessarily high bandwidth requirements, the performance of the entire system can be enhanced in terms of playback delay and SVC video quality by up to 20%. At the same time, content providers are able to reduce up to 60 of their server costs, compared to the classical QoS-based approach.

Multimedia applications have become an essential part of our daily lives, and their use is flourishing day by day. In Wireless Multimedia Sensor Networks (WMSNs), large volume of multimedia data is transmitted through the network. If the traffic load is greater than the available capacity of the sensor network, congestion occurs and it causes buffer overflow, packet drop, deterioration of network throughput and effect quality of service (QoS). The limited bandwidth should be adjusted for the packets with different priorities precisely. To emphasize the problem, the paper provides A New Approach for Traffic Management in Wireless Multimedia Sensor Network. According to the paper, sensor nodes in one collision area can adjust their sending states adaptively based on packet priority. To improve the received video quality in base stations, proposed technique keeps the I-frames and ignores the other less important frame types of compressed video. Also retransmissions of lost packets are provided with different priorities for enhancing the system performance. The simulations show that the bandwidth adjustment can protect the packets with high priority effectively, and the performance of the retransmission is well positioned to meet the requirement of the multimedia applications. MATLAB is used for simulation purpose.

This paper describes a three-screen television system using a block recovery rate (BRR)-based unequal error protection (UEP). The proposed in-home wireless network uses scalable video coding (SVC) and UEP with forward error correction (FEC) for maximizing the quality of service (QoS) over error-prone wireless networks. For efficient FEC packet assignment, this paper proposes a simple and efficient performance metric, a BRR which is defined as a recovery rate of temporal and quality layer from FEC assignment by analyzing the hierarchical prediction structure including the current packet loss. It also explains the SVC layer switching scheme according to network conditions such as packet loss rate (PLR) and available bandwidth (ABW). In the experiments conducted, gains in video quality with the proposed UEP scheme vary from 1 to 3 dB in Y-peak signal-to-noise ratio (PSNR) with corresponding subjective video quality improvements.

Multimedia applications are routed from source nodes via neighboring nodes to a sink in Wireless Multimedia Sensor Networks (WMSNs). However, the existing routing protocols in the WMSNs do not fulfill the quality of service (QoS) constraint yet for multimedia transmission. The WMSNs have specific characteristics such as multi-hop way, limited node capability and varying bandwidth, which can affect the quality of service of multimedia applications during the routing process. The best quality of service (QoS) is required by the multimedia applications consisting of high throughput, low end-to-end delay and low packet drop rate. A routing optimization based on the cross-layer design is proposed for multimedia transmission in the WMSNs. The cross-layer design utilizes H.264/Scalable Video Coding (SVC) and IEEE 802.11e Enhanced Distributed Channel Access (EDCA) and has been implemented in Network Simulator 2 (NS2) simulation. The NS2 simulation indicates that the H.264/SVC video application has an average delay of 0.02 sec. Thus, the routing optimization has accomplished to deliver multimedia application successfully via Wireless Multimedia Sensor Network (WMSN).

Recent advances in wireless broadband networks and video coding techniques have led the rapid growth of wireless video services. In this chapter, we present a comprehensive study on the transmission of scalable video over wireless local area networks (WLAN). We analyze first the mechanisms and principles of the emerging scalable video coding (SVC) standard. We then introduce the IEEE 802.11 standards for WLAN and related quality of service (QoS) issues. We present some studies of SVC over WLAN using cross-layer design techniques. We aim to exploit the unique characteristics of the scalable video coding technology to enhance personalized experience and to improve system performance in a wireless transmission system. Examples and analyses are given to demonstrate system performances.

<p>It is now well known that employing channel knowledge based on signaling techniques in wireless mobile ad-hoc networks (MANET) system can yield large improvements in almost all performance metric. Here we proposed the adaptive scheduling, in which the work done is based upon the bandwidth information of channel to provide better quality of service (‘QoS’) to the cell-edge mobile stations. Channel information is critical based on which scheduling is carried out. The bandwidth channel information contains estimation delay, the pilot channel noise and pilot contamination. Afterwards, Zero Forcing precoding methodology has applied for removing the interference at user nodes, destination nodes and gateway side. By extending the characteristics of ZF, the modified Zero Forcing (MZF) has proposed to achieve higher throughput rate and higher spectrum efficiency. The achievable-rates of the ZF and MZF has derived under the comprehensive model of imperfect bandwidth information.</p>

Multimedia applications are much more sensitive to the quality of service(QoS) than text-based ones due to their data continuity. In order to provide a fast moving mobile host(MH) supporting multimedia applications with a consistent QoS, an efficient call admission mechanism is in need. This paper proposes the 2-level call admission control(2LCAC) scheme based on a call admission scheme using the priority to guarantee the consistent QoS for mobile multimedia applications. The 2LCAC consists of the basic call admission and advanced call admission; the former determines call admission based on bandwidth available in each cell and the latter determines call admission by utilizing delay tolerance time(DTT), priority queue(PQueue), and upgrade queue(UpQueue) algorithms. In order to evaluate the performance of our scheme, we measure the metrics such as the blocking probability of new calls, dropping probability of handoff calls, and bandwidth utilization. The result shows that the performance of our scheme is superior to that of existing schemes such as complete sharing policy(CSP), guard channel policy(GCP) and adaptive guard channel policy(AGCP).

International audience

This research addressed the issue of improving the quality of service for the control system of mobile radio networks. The analysis of the forecasting sphere concerning the methods of service quality of mobile radio networks for special purposes, in particular, forecasting the time of congestion of data transmission routes is carried out. It is found that these methods are used in wired and computer networks operating at the network and data link levels. The basic parameters of the protocols of the channel and network layers of mobile radio networks are highlighted. Forecasting methods are analyzed: temporal extrapolation, causality, expert, and the main disadvantages are indicated. A model of a control system for mobile radio networks with a forecasting subsystem is shown. The features of mobile radio networks, which form the requirements for routing methods, are described. A lot of requirements have been put forward for the model of a control system for mobile radio networks. The structure of a model of a control system for mobile radio networks with an improved forecasting subsystem is proposed. On the basis of genetic algorithms, the tasks that arise in the process of identification, training and forecasting in the forecasting subsystem are solved. The operation of the processes consists in building a base of rules aimed at identifying significant dependencies in a time series based on the use of a genetic algorithm. It is based on the use of evolutionary principles to find the optimal solution. Application of the proposed model will allow real-time identification and will significantly improve the quality of service for mobile radio networks. It will increase the speed and volume of data processed during training, improve the quality and reliability of predicting changes in data transmission routes