无线网络视频传输差错控制技术研究

随着多媒体、网络技术以及移动通信的发展,视频通信的应用成了必然的趋势。传输的视频需要进行压缩,冗余信息的丢失使视频数据在传输中抵抗信道误码的能力变得十分脆弱。不幸的是,无线网络信道中,误码的产生、数据的丢失总是难以避免。当网络拥塞时更容易造成突发性的分组丢失现象,引起图像质量严重下降,必须采用有效的差错控制技术进行处理。本文提出并实现了一种适合于无线网络环境下视频传输的差错控制方法,它包括一个基于丢包检测的流量自适应算法,编码端的宏块重排序的算法和解码端的自适应的差错掩藏算法。实验结果表明,在无线网络环境下,采用本文提出的差错控制方法能够有效提升视频传输的质量。     

一种无线实时流媒体增强型自适应FEC控制机制

无线网络动态的信道特性和带宽有限等特点,使得在无线环境下为流媒体应用提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的增强型自适应前向纠错控制策略,以提高接收方的播放质量。该策略采用跨层设计的方法,根据当前的网络状态,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

Header Detection to Improve Multimedia Quality Over Wireless Networks

Wireless multimedia studies have revealed that forward error correction (FEC) on corrupted packets yields better bandwidth utilization and lower delay than retransmissions. To facilitate FEC-based recovery, corrupted packets should not be dropped so that maximum number of packets is relayed to a wireless receiver's FEC decoder. Previous studies proposed to mitigate wireless packet drops by a partial checksum that ignored payload errors. Such schemes require modifications to both transmitters and receivers, and incur packet-losses due to header errors. In this paper, we introduce a receiver-based scheme which uses the history of active multimedia sessions to detect transmitted values of corrupted packet headers, thereby improving wireless multimedia throughput. Header detection is posed as the decision-theoretic problem of multihypothesis detection of known parameters in noise. Performance of the proposed scheme is evaluated using trace-driven video simulations on an 802.11b local area network. We show that header detection with application layer FEC provides significant throughput and video quality improvements over the conventional UDP/IP/802.11 protocol stack     

Cross-Layer Error Control for Multimedia Streaming in Wireless/Wireline Packet Networks

In this paper, we propose a cross-layer error control framework for robust and low delay multimedia streaming in tandem-connected IEEE 802.11 wireless LANs and the Internet. For this network configuration, we model the end-to-end delay and packet loss rate as a function of the automatic repeat request (ARQ) and forward error correction (FEC) error control mechanisms that are employed at the application and wireless link layers. The analytical model is used as the basis of a delay-constrained error control algorithm that adapts the protection level at the application and link layers so that the end-to-end packet loss rate is minimized. With extensive simulations, we validate the efficiency of the proposed cross-layer error control methodology for delay-sensitive pre-compressed video streaming.      

无线上行视频流的跨层容错传输方法

如何通过资源受限的移动通信终端提升无线上行视频流的抗误性能是亟待解决的重要问题。通过不同通信层次的联合调度,提出了一种跨层容错传输方案。移动通信终端的网络层代理首先利用容错包调度为视频流的延时约束帧集合提供重要性分类,随后该终端的链路层代理利用无线链路单元的优先级调度实现选择性重传。在调度延时与传输带宽限制下,跨层容错传输能够将突发错误转移到延时约束帧集合的低优先级视频数据中,从而在突发易错传输环境中实现了无线链路单元粒度的渐进式传输和平稳退化。     

基于MPEG4的自适应实时流媒体传输

论文阐述了在互联网环境下进行实时流媒体传输中存在的一些难点,并且给出了一个基于MPEG4的实时流媒体传输系统。并且希望能充分利用网络带宽,并且尽力减少媒体流由于网络时延和丢包所产生的影响。同时也关注由于丢包而引起的回放质量的下降问题。     

Providing adaptive QoS to layered video over wireless local area networks through real-time retry limit adaptation

Robust streaming of video over 802.11 wireless LANs (WLANs) poses many challenges, including coping with packets losses caused by network buffer overflow or link erasures. In this paper, we propose a novel error protection method that can provide adaptive quality-of-service (QoS) to layered coded video by utilizing priority queueing at the network layer and retry-limit adaptation at the link layer. The design of our method is motivated by the observation that the retry limit settings of the MAC layer can be optimized in such a way that the overall packet losses that are caused by either link erasure or buffer overflow are minimized. We developed a real-time retry limit adaptation algorithm to trace the optimal retry limit for both the single-queue (or single-layer) and multiqueue (or multilayer) cases. The video layers are unequally protected over the wireless link by the MAC with different retry limits. In our proposed transmission framework, these retry limits are dynamically adapted depending on the wireless channel conditions and traffic characteristics. Furthermore, the proposed priority queueing discipline is enhanced with packet filtering and purging functionalities that can significantly save bandwidth by discarding obsolete or un-decodable packets from the buffer. Simulations show that the proposed cross-layer protection mechanism can significantly improve the received video quality.     

Multiple TFRC Connections Based Rate Control for Wireless Networks

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control , in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TCP friendly rate control (TFRC), assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. In this paper, we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over 1$times$RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach.     

Adaptive hybrid error correction model for video streaming over wireless networks

The Hybrid ARQ (HARQ) mechanism is the well-known error packet recovery solution composed of the Automation Repeat reQuest (ARQ) mechanism and the Forward Error Correction (FEC) mechanism. However, the HARQ mechanism neither retransmits the packet to the receiver in time when the packet cannot be recovered by the FEC scheme nor dynamically adjusts the number of FEC redundant packets according to network conditions. In this paper, the Adaptive Hybrid Error Correction Model (AHECM) is proposed to improve the HARQ mechanism. The AHECM can limit the packet retransmission delay to the most tolerable end-to-end delay. Besides, the AHECM can find the appropriate FEC parameter to avoid network congestion and reduce the number of FEC redundant packets by predicting the effective packet loss rate. Meanwhile, when the end-to-end delay requirement can be met, the AHECM will only retransmit the necessary number of redundant FEC packets to receiver in comparison with legacy HARQ mechanisms. Furthermore, the AHECM can use an Unequal Error Protection to protect important multimedia frames against channel errors of wireless networks. Besides, the AHECM uses the Markov model to estimate the burst bit error condition over wireless networks. The AHECM is evaluated by several metrics such as the effective packet loss rate, the error recovery efficiency, the decodable frame rate, and the peak signal to noise ratio to verify the efficiency in delivering video streaming over wireless networks.     

A robust method for soft IP handover

Multimedia streaming over wireless networks - often called mobile multimedia streaming lets users access music, movie, and news services at any time, regardless of location. Given that multimedia streaming is a key goal of third-generation and future wireless networks, vendors will soon deploy streaming clients in advanced mobile terminals. Current mobile terminals, however, fail to adequately support mobile multimedia communication because wireless networks have high packet-loss rates. To eliminate packet loss during handover, we use a packet path diversity scheme and an end-to-end bicasting mechanism that enables soft IP handover. To offset wireless errors, we use a forward error correction (FEC) scheme and embed it in the bicasting mechanism. Our bicasting method encodes the data stream and then splits it, providing more effective diversity than general bicasting, which sends the same data down both paths.' To support our method, we propose the mobile multimedia streaming protocol (MMSP), a new transport-layer protocol that supports multihoming and bicasting in combination with FEC.     

A portable medical system using real-time streaming transport over 3G wireless networks

The paper proposes architecture for the software implementation of a multimedia, tele-medicine system. It considers video streaming from both video servers in hospitals and webcams localized to patients. It also considers transmission of vital bio-signs, such as heart rates and blood pressure, etc. All these data are transmitted over a 3G-wireless communication system to various client devices (hand-held devices, such as PDAs) used by physicians and nurses. Our video codec is a software implementation of the MPEG-4 standard, compression rate at about 1/24 sizes suited for available transmitting bandwidth. Moreover, our design, which also integrates the processing of heart sounds, supports a 44.1 KHz sampling rate and a 16-bit representation required about 11 kbps bandwidth. At the same time, in the streaming process, we propose a congestion control scheme to reduce packet losses.     

Distributed Collaboration for Enhanced Sender-Driven Video Streaming

We propose a sender-driven system for adaptive streaming from multiple servers to a single receiver over separate network paths. The servers employ information in receiver feedbacks to estimate the available bandwidth on the paths and then compute appropriate transmission schedules for streaming media packets to the receiver based on the bandwidth estimates. An optimization framework is proposed that enables the senders to compute their transmission schedules in a distributed way, and yet to dynamically coordinate them over time such that the resulting video quality at the receiver is maximized. To reduce the computational complexity of the optimization framework an alternative technique based on packet classification is proposed. The substantial reduction in online complexity due to the resulting packet partitioning makes the technique suitable for practical implementations of adaptive and efficient distributed streaming systems. Simulations with Internet network traces demonstrate that the proposed solution adapts effectively to bandwidth variations and packet loss. They show that the proposed streaming framework provides superior performance over a conventional distortion-agnostic scheme that performs proportional packet scheduling on the network paths according to their respective bandwidth values.     

一种改进的瑞利小波算法无线传感器网络流媒体传输研究

针对无线传感器网络中带宽和能量受限、误码率高、信道不稳定等因素严重影响了实时流媒体传输的问题,采用瑞利小波模型模拟无线传感器网络流媒体通信,并给出概率分布和突发特性的分析模型,基于Kal-man滤波器实时预测网络带宽,自适应地在SCTP与PRSCTP之间进行切换。仿真实验表明,瑞利小波模型能够准确地描述实时流媒体通信流,Kalman滤波器可以准确地预测实时网络带宽,而且基于带宽预测的流媒体传输技术与原有的技术相比在分组成功投递率、端到端时延和吞吐率上均具有良好的性能。     

一种自适应的视频流化前向纠错算法

网络视频应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰.相关研究表明:在多数情况下,动态变化的网络带宽和丢包率是影响视频流化质量的关键因素.因此,为了保证视频质量,可以采用前向纠错(forward error correction,简称FEC)编码来提高视频数据传输的可靠性;同时,为了适应网络状态的变化,发送端可以调节视频数据的发送速率,并在视频源数据与FEC数据之间合理分配网络传输带宽.首先通过对视频流结构的分析,在充分考虑帧之间的依赖关系和帧类型的基础上提出了一种帧的解码模型.在此基础上,建立了用于在视频源数据和FEC数据之间分配网络带宽资源的优化算法.实验表明,该模型可以有效地适应网络状态的变化,并通过优化分配网络带宽资源来使接收端获得最大的可播放帧率.     

Kalman滤波的自适应链路层FEC控制策略

无线网络动态的信道特性、高误码率和带宽有限等特点,使得在无线环境下为实时流媒体传输提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的自适应链路层FEC控制策略,以显著提高接收方的播放质量。该策略采用跨层设计的方法,基于Kalman滤波器预测当前的网络状态,考虑物理带宽限制和GOP可解码帧数的特性自适应地调整FEC参数N;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。     

基于小波EBCOT的图像IP网络传输控制策略

为了使当前“尽力而为”的网络提供视频流服务时满足QoS要求，文章提出一种基于小波EBCOT的图像IP网络传输控制策略．通过采用基于小波EBCOT的渐进可分级编码方法，对压缩后的比特流按其重要性分层打包传输，同时根据对当前网络可用带宽的估计及信道状态的判断，区分网络拥塞及不可靠传输两种不同情况进行自适应不等重丢包保护AUPLP．软件仿真表明，该文算法可大大增强小波EBCOT编码后数据的抗误码能力，在发生数据拥塞时有助于缓解网络的过负载状况，在发生不可靠传输时接收端解码图像能平均提高1．2dB的PSNR。     

双路信道无线视频传输系统设计

为了在限制带宽和高误码率的公用移动通信网络上获得高质量的实时视频传输,提出了一种基于双路增强型数据速率GSM演进技术(EDGE)信道的无线视频传输原型系统。系统以高性能的数字信号处理器(DSP)为核心设计无线视频终端,采用H.264编码,利用双路EDGE传输信道,提供较高的传输带宽。运用多缓冲的发送机制和差错控制策略,将视频流发送到视频服务器。实验结果表明,CIF格式的图像传输能达到基本实时的要求。     

Video Streaming Over In-Home Power Line Networks

The deployment of power line communication technology for broadband video streaming remains a challenge because power lines are not originally designed for signal transmission. Scalable video is a viable approach that can cope with the bandwidth fluctuation of power line communication networks provided that the bandwidth information is available. In this paper we first investigate how the interference caused by electrical appliances or power supplies affects the power line channel bandwidth and packet transmission. Then we take the obtained characteristics of in-home power line network into account in the design of a simple but effective heuristic-based application-layer bandwidth estimation scheme, for which the cutoff rate is estimated from the packet size and the physical-layer data rates. Experimental results show that the proposed approach can effectively combat the noise interference and deliver robust video streaming over power line.      

Eliminating the effect of freezing frames on user perceptive by using a time interleaving technique

Streaming video over a wireless network faces several challenges such as high packet error rates, bandwidth variations, and delays, which could have negative effects on the video streaming and the viewer will perceive a frozen picture for certain durations due to loss of frames. In this study, we propose a Time Interleaving Robust Streaming (TIRS) technique to significantly reduce the frozen video problem and provide a satisfactory quality for the mobile viewer. This is done by reordering the streaming video frames as groups of even and odd frames. The objective of streaming the video in this way is to avoid the losses of a sequence of neighbouring frames in case of a long sequence interruption. We evaluate our approach by using a user panel and mean opinion score (MOS) measurements; where the users observe three levels of frame losses. The results show that our technique significantly improves the smoothness of the video on the mobile device in the presence of frame losses, while the transmitted data are only increased by almost 9% (due to reduced time locality).     

Wireless network interface energy consumption

With the proliferation of mobile streaming multimedia, available battery capacity constrains the end-user experience. Since streaming applications are expected to be long running, wireless network interface card's (WNIC) energy consumption is particularly an acute problem. In this work, we explore various mechanisms to conserve client WNIC energy consumption for popular streaming formats such as Microsoft Windows media, Real and Apple Quicktime. First, we investigate the WNIC energy consumption characteristics for these popular multimedia streaming formats under varying stream bandwidth and network loss rates. We show that even for a high bandwidth 2000 kbps stream, the WNIC unnecessarily spent over 56% of the time in idle state; illustrating the potential for significant energy savings.Based on these observations, we explore two mechanisms to conserve the client WNIC energy consumption. First we show the limitations of IEEE 802.11 power saving mode for multimedia streams. Without an understanding of the stream requirements, these scheduled rendezvous mechanisms do not offer any energy savings for multimedia streams over 56 kbps. We also develop history-based client-side strategies to reduce the energy consumed by transitioning the WNICs to a lower power consuming sleep state. We show that streams optimized for 28.8 kbps can save over 80% in energy consumption with 2% data loss. A high bandwidth stream (768 kbps) can still save 57% in energy consumption with less than 0.3% data loss. We also show that Real and Quicktime packets are harder to predict at the network level without understanding the packet semantics. As the amount of cross traffic generated by other clients that share the same wireless segment increases, the potential energy savings from our client side policies deteriorate further. Our work enables multimedia proxy and server developers to suitably customize the stream to lower client energy consumption.