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

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.     

Radio Link Buffer Management and Scheduling for Wireless Video Streaming

In this paper we compare strategies for joint radio link buffer management and scheduling for wireless video streaming. Based on previous work in this area [8], we search for an optimal combination of scheduler and drop strategy for different end-to-end streaming options including timestamp-based streaming and ahead-of-time streaming, both with variable initial playout delay. We will show that a performance gain versus the two best drop strategies in Liebl et al. [8], i.e. drop the HOL packet or drop the packet with the lowest priority starting from HOL, is possible: Provided that some basic side-information on the structure of the incoming video stream is available, a more sophisticated drop strategy removes packets from an HOL group of packets in such a way that the temporal dependencies usually present in video streams are not violated. This advanced buffer management scheme yields significant improvements for almost all investigated scheduling algorithms and streaming options. In addition, we will demonstrate the importance of fairness among users when selecting a suitable scheduler, especially if ahead-of-time streaming is to be applied: Given a reasonable initial playout delay at the streaming media client, both the overall achievable quality averaged over all users, as well as the individual quality of users with bad channel conditions can be increased significantly by trading off fairness with maximum throughput of the system.     

无线链路多用户视频流传输QoS保证：延时限制及优先级丢包

To provide a certain level of Quality of Service (QoS) guarantees for multiuser wire-less downlink video streaming transmissions, we propose a multiuser scheduling scheme for QoS guarantees. It is based on the classic Queue-Length-Based (QLB)-rate maximum scheduling algorithm and integrated with the delay constraint and the packet priority drop. We use the large deviation principle and the effective capacity theory to construct a new analysis model to find each user’s queue leng-th threshold (delay constraint) violation prob-ability. This probability corresponds to the upper bound of the packet drop probability, which indicates a certain level of statistical QoS guarantees. Then, we utilize the priority information of video packets and introduce the packet priority drop to further improve the quality perceived by each user. The simu-lation results show that the average Peak Signal to Noise Ratio (PSNR) value of the priority drop is 0.8 higher than that of the non-priority drop and the PSNR value of the most badly damaged video frame in the priority drop is on an average 4 higher than that of the non-priority drop.     

Priority-Aware Packet Pre-marking for DiffServ Architecture Based on H.264/SVC Video Stream Structure

The H264/SVC codec allows for generation of hierarchical video streams. In the stream of this type video data belonging to different layers have different priority depending on their importance to the quality of the video and the decoding process. This creates new demands on the mechanisms of packet marking, and thus new challenges for the policy guaranteeing QoS parameters, such as those defined in the DiffServ architecture. Therefore, mechanisms of the traffic engineering used in the DiffServ network should, as far as possible, take into account internal distribution of priorities inside video streams. This may be achieved by implementing an appropriate method for packet pre-marking. The paper describes the Weighted Priority Pre-marking (WPP) algorithm for priority-aware SVC video streaming over a DiffServ network. Our solution takes into account the relative importance of the Network Abstraction Layer Units. It also does not require any changes in the implementation of the DiffServ marker algorithm. The results presented confirm that video transmission in the DiffServ domain, based on the WPP packet pre-marking, can provide better perceived video quality than the standard (best effort) streaming of multi-layered SVC video. In addition, a comparison with the transmission of the same video content encoded with the H264/AVC codec also points to the superiority of our proposed method.

     

OFDM系统中视频传输的调度算法研究

针对OFDM系统,提出了一种基于视频内容的跨层调度方案。该方案采用了基于生存期的包排序策略,根据视频包的生存期大小进行排序,保证视频包能在生存期内发送到接收端,减少丢包数目。同时,采用跨层设计的思想,综合考虑了信道状况和视频包特性,如视频包生存期、重要性、解码器采用的错误掩藏方法,提出改进的比例公平调度算法,不仅有效地利用了多用户分集来进一步提高数据吞吐量,也充分考虑了视频的重要性和时延约束。实验结果表明,采用内容感知的跨层调度算法,解码端的视频质量得到有效提高,从而可以提高主观感知质量。     

An adaptive retransmission‐based multipath transmission mechanism for conversational video

Conversational video service is characterized by high bandwidth demand and low delay requirement. Bandwidth and transmission schemes play an important role in providing high‐quality delivery service for point‐to‐point conversational video service. Multipath transmission is regarded as an effective way to aggregate bandwidth. Transmission schemes need to ensure the strict time relation between information entities and to alleviate the negative impact of packet loss on video quality. To achieve this, existing transmission schemes may incur either a large delay or a large amount of duplicated packets that are not suitable for conversational video service. In this paper, we propose an adaptive retransmission mechanism–based multipath transmission (MT‐AR) for conversational video service delivery. MT‐AR takes advantage of historical reception experience to timely detect packet loss with a certain degree of misjudgement. Receiver requests sender to retransmit the lost packet if the lost packet benefits the decoding. Adaptive playout speed adjustment and alternative path retransmission cooperatively optimize the performance of retransmission. Receiver slightly extends playout speed to reserve time for retransmission and accelerates playout speed to alleviate negative impact of cumulative extension. Multiple paths support to conduct retransmission on an optimal path selected from alternative paths to avoid continuous congestion or error on the original path. Finally, we conduct extensive tests to evaluate the performance of MT‐AR. Experimental results show that MT‐AR can effectively improve the quality of experience of conversational video service by retransmission.     

Hierarchical constellation for multi-resolution data transmission over block fading channels

We propose a new technique for multi-resolution video/image data transmission over block fading channels. The proposed scheme uses an adaptive scheduling protocol employing a retransmission strategy in conjunction with a hierarchical signal constellation (known also as nonuniform, asymmetric, multi-resolution constellation) to give different transmission priorities to different resolution levels. Transmission priorities are given in terms of average packet loss rate as well as average throughput. Basically, according to the transmission scheduling and channel state (acknowledgment signal) of the previous transmission, it dynamically selects packets from different resolution levels to transmit for the current transmission. The bits from the selected packets are assigned to different hierarchies of a hierarchical 4/16-quadrature amplitude modulation to transmit them with different error protections. The selection of packets for transmission and the assignment of these selected packets to different hierarchies of the hierarchical constellation are referred to as the scheduling protocol in our proposed scheme. We model this protocol by a finite state first order Markov chain and obtain the packet loss rate and the packet transmission rate over Nakagami-m block fading channel in closed-form. Some selected numerical results show that the proposed scheme can control the relative packet loss rate and the packet transmission rate of different resolution levels by varying the priority parameter (or equivalently, the asymmetry) of the hierarchical constellation and the maximum number of allowed retransmissions.     

A lightweight network state estimation mechanism in ARQ-based wireless networks

Most multimedia applications implement some kind of packet losses analysis mechanism to trigger self-adaptation actions. However, error recovery mechanisms such as ARQ variants avoid packet losses at IP level. Then, there appears an impact into the delay that may indeed result on application-level losses due to samples arriving later than the scheduled playout time. In most cases such degradations are not detected until they are so severe than even losses at IP level appear. We propose a lightweight method that achieves a finer grain estimation. We prove mathematically the capability of modified versions of statistics of delay to predict the error ratio of the wireless link and the load of the wired backhaul. After deducing also simplified heuristics for proposed method, we analyze their estimation capabilities and provide guidance for selecting appropriate parameters. Finally, we test and adjust the algorithm to an specific scenario including mobile video streaming and VoIP calls.     

突发大时延下的IP语音缓冲算法

当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。     

Optimizing the ARQ performance in downlink packet data systems with scheduling

Third generation wireless systems typically employ adaptive coding and modulation, scheduling, and Hybrid Automatic Repeat reQuest (HARQ) techniques to provide high-speed packet data service on the downlink. Two main considerations in designing such a system are algorithms for the selection of coding and modulation schemes based on the channel quality of the link and algorithms for the selection of the user to whom a particular slot is assigned. We propose a systematic approach to optimize the mapping between signal-to-interference-and-noise ratio (SINR) and modulation and coding scheme (MCS) to maximize the throughput by taking into account the type of HARQ scheme employed. We also propose to incorporate frame error rate (FER) and retransmission information as a part of the scheduling decision. The proposed scheduler ranking methods based on using an effective rate rather than the instantaneous rate provide natural priority to retransmissions over new transmissions, and priority to users with better channel quality. Extensive simulation results comparing performance of the proposed methods to conventional methods are presented.     

Random linear network coding with ladder-shaped global coding matrix for robust video transmission

Robust video multicast in erasure networks using network coding (NC) to reduce the impact of packet loss is studied in this paper. In our proposed solution, random linear network coding (RLNC) is adopted at intermediate nodes of the network. RLNC linearly combines a group of packets by randomly selecting weighting coefficients on a finite field, and the loss of an RLNC-coded packet is equivalent to the loss of one constraint in a linear system of equations required for RLNC decoding. Unless the global coding coefficient matrix, or simply called the global coding matrix (GCM), is of full rank, a receive node cannot reconstruct all source packets. To address this rank deficiency problem, we propose to construct a special-structured GCM, called the ladder-shaped GCM (LGCM), for layered H.264/SVC (scalable video coding) video multicast. The ladder shape of the sparse coding matrix is maintained throughout the RLNC process to achieve two objectives: (1) to enable partial decoding of a block; and (2) to provide unequal erasure protection for H.264/SVC priority layers. Furthermore, quality degradation is minimized by optimizing the amount of redundancy assigned to each layer, and graceful quality degradation is achieved by error concealment (EC). Simulation results are given to demonstrate the superior performance of the proposed RLNC–LGCM scheme over the traditional RLNC with a generalGCM.     

Joint packet scheduling and resource allocation with quality fairness for wireless VoD system

This paper investigates the problem of multiuser packet scheduling and resource allocation for video transmission over downlink OFDMA networks. A cross-layer approach is proposed to maximize the received video quality under the video quality fairness constraint. Unlike the previous methods in which the objective index is estimated the video quality in the unit of bit, the proposed algorithm develops the objective index in unit of packet, which is more fit for video transmission. In order to solve the optimization problem, a suboptimal algorithm of joint packet scheduling and resource allocation is proposed. The algorithm is compatible with the emerging wireless standards, such as IEEE 802.16. The simulation results show that the proposed method outperforms the conventional resource allocation schemes in terms of received video qualities and quality fairness.     

Efficient scheduling discipline for Hierarchical Diff‐EDF

Packet networks are currently enabling the integration of traffic with a wide range of characteristics that extend from video traffic with stringent quality of service (QoS) requirements to the best‐effort traffic requiring no guarantees. QoS guarantees can be provided in conventional packet networks by the use of proper packet‐scheduling algorithms. As a computer revolution, many scheduling algorithms have been proposed to provide different schemes of QoS guarantees, with Earliest Deadline First (EDF) as the most popular one. With EDF scheduling, all flows receive the same miss rate regardless of their traffic characteristics and deadlines. This makes the standard EDF algorithm unsuitable for situations in which the different flows have different miss rate requirements since in order to meet all miss rate requirements it is necessary to limit admissions so as to satisfy the flow with the most stringent miss rate requirements. In this paper, we propose a new priority assignment scheduling algorithm, Hierarchal Diff‐EDF (Differentiate Earliest Deadline First), which can meet the real‐time needs of these applications while continuing to provide best‐effort service to non‐real time traffic. The Hierarchal Diff‐EDF features a feedback control mechanism that detects overload conditions and modifies packet priority assignments accordingly. Copyright © 2007 John Wiley & Sons, Ltd.     

A distributed utility‐based scheduling for peer‐to‐peer video streaming over wireless networks

Interactive multimedia applications such as peer‐to‐peer (P2P) video services over the Internet have gained increasing popularity during the past few years. However, the adopted Internet‐based P2P overlay network architecture hides the underlying network topology, assuming that channel quality is always in perfect condition. Because of the time‐varying nature of wireless channels, this hardly meets the user‐perceived video quality requirement when used in wireless environments. Considering the tightly coupled relationship between P2P overlay networks and the underlying networks, we propose a distributed utility‐based scheduling algorithm on the basis of a quality‐driven cross‐layer design framework to jointly optimize the parameters of different network layers to achieve highly improved video quality for P2P video streaming services in wireless networks. In this paper, the quality‐driven P2P scheduling algorithm is formulated into a distributed utility‐based distortion‐delay optimization problem, where the expected video distortion is minimized under the constraint of a given packet playback deadline to select the optimal combination of system parameters residing in different network layers. Specifically, encoding behaviors, network congestion, Automatic Repeat Request/Query (ARQ), and modulation and coding are jointly considered. Then, we provide the algorithmic solution to the formulated problem. The distributed optimization running on each peer node adopted in the proposed scheduling algorithm greatly reduces the computational intensity. Extensive experimental results also demonstrate 4–14 dB quality enhancement in terms of peak signal‐to‐noise ratio by using the proposed scheduling algorithm. Copyright © 2015 John Wiley & Sons, Ltd.     

Efficient Traffic Scheduling for Real Time VBR MPEG Video Transmission Over DOCSIS-Based HFC Networks

Data-over-cable service interface specifications (DOCSIS), the de facto standard in the cable industry, defines a scheduling service called real-time polling service (rtPS) to provision quality of service (QoS) transmission of real-time variable bit rate (VBR) videos. However, the rtPS service intrinsically has high latency, which makes it not applicable to real-time traffic transport. In this paper, we present a novel traffic scheduling algorithm for hybrid fiber coax (HFC) networks based on DOCSIS that aims to provide QoS for real-time VBR video transmissions. The novel characteristics of this algorithm, as compared to those described in published literatures, include 1) it predicts the bandwidth requirements for future traffic using a novel traffic predictor designed to provide simple yet accurate online prediction; and 2) it takes the attributes of physical (PHY) layer, media access control (MAC) layer and application layer into consideration. In addition, the proposed traffic scheduling algorithm is completely compatible with the DOCSIS specification and does not require any protocol changes. We analyze the performance of the proposed traffic predictor and traffic scheduling algorithm using real-life MPEG video traces. Simulation results indicate that 1) the proposed traffic predictor significantly outperforms previously published techniques with respect to the prediction error and 2) Compared with several existing scheduling algorithms, the proposed traffic scheduling algorithm surpasses other mechanisms in terms of channel utilization, buffer usage, packet delay, and packet loss rate.     

Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications

An end-to-end packet delay in the Internet is an important performance parameter, because it heavily affects the quality of real-time applications. In the current Internet, however, because the packet transmission qualities (e.g., transmission delays, jitters, packet losses) may vary dynamically, it is not easy to handle a real-time traffic. In UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at a client host to compensate for variable delays. The issue of playout control has been studied by some previous works, and several algorithms controlling the playout buffer have been proposed. These studies have controlled the network parameters (e.g., packet loss ratio and playout delay), not considered the quality perceived by users. In this paper, we first clarify the relationship between Mean Opinion Score (MOS) of played audio and network parameters (e.g., packet loss, packet transmission delay, transmission rate). Next, utilizing the MOS function, we propose a new playout buffer algorithm considering user's perceived quality of real-time applications. Our simulation and implementation tests show that it can enhance the perceived quality, compared with existing algorithms.     

Bluetooth Dynamic Scheduling and Interference Mitigation

Bluetooth is a cable replacement technology for Wireless Personal Area Networks. It is designed to support a wide variety of applications such as voice, streamed audio and video, web browsing, printing, and file sharing, each imposing a number of quality of service constraints including packet loss, latency, delay variation, and throughput. In addition to QOS support, another challenge for Bluetooth stems from having to share the 2.4 GHz ISM band with other wireless devices such as IEEE 802.11. The main goal of this paper is to investigate the use of a dynamic scheduling algorithm that guarantees QoS while reducing the impact of interference. We propose a mapping between some common QoS parameters such as latency and bit rate and the parameters used in the algorithm. We study the algorithm's performance and obtain simulation results for selected scenarios and configurations of interest.     

Combined Connection Admission Control and Packet Transmission Scheduling for Mobile Internet Services

Mobile Internet access is expected to be the most popular communication service in the near future. In this paper, we investigate radio resource management for mobile Internet multimedia systems that use the orthogonal frequency division multiple access and adopt the adaptive modulation and coding technique. It is assumed that real-time (RT) service such as streaming video and best-effort (BE) services such as file transfer protocol and hypertext transfer protocol coexist in the systems. We suggest two levels of radio resource management schemes: the connection admission control (CAC) scheme at the first level and the packet transmission scheduler at the second level. The proposed scheduler does not assign higher priority to RT packets over BE packets unconditionally. Instead, only the RT packets that are close to the deadline are given higher priority. Therefore, the performance of BE services is improved at the cost of RT services. To control the performance degradation in RT services within an acceptable level, the CAC algorithm functions as a congestion controller. The combined effects of the proposed CAC and packet scheduling by using the cross-layer simulation that covers from the physical layer to the Internet application layer are evaluated. The numerical results show that the proposed schemes greatly improve the performance of BE services while maintaining the quality of video service at an acceptable level.     

Adaptive error-resilience transcoding using prioritized intra-refresh for video multicast over wireless networks

In this paper, we propose a two-pass error-resilience transcoding scheme based on adaptive intra-refresh for inserting error-resilience features to a compressed video at the intermediate transcoder of a three-tier streaming system. The proposed transcoder adaptively adjusts the intra-refresh rate according to the video content and the channel's packet-loss rate to protect the most important macroblocks against packet loss. In this work, we consider the problem of multicast of video to multiple clients having disparate channel-loss profiles. We propose a MINMAX loss rate estimation scheme to determine a single intra-refresh rate for all the clients in a multicast group. For the scenario that a quality variation constraint is imposed on the users, we also propose a grouping method to partition a multicast group of heterogeneous users into a minimal number of subgroups to minimize the channel bandwidth consumption while meeting the quality variation constraint. Experimental results show that the proposed method can effectively mitigate the error propagation due to packet loss as well as achieve fairness among clients in a multicast.