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.     

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.     

AVP: A Highly Efficient Transport Protocol for Low Bit Rate Multimedia Communications

Utilization of Internet communications in distance learning, distributed simulation, and distributed work groups involves multimedia transmission of animation, voice and video clips. Highly compressed audio-video data protocols are required for efficient Internet multimedia communications. Addressing this requirement, a new transport protocol called Audio-Video Protocol (AVP) for highly efficient multimedia communications on the Internet is presented. While providing similar real-time delivery functions as Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP), AVP adopts a novel audio-based synchronization scheme. This synchronization scheme has two advantages. One is the overhead reduction through eliminating the timestamp in each transmitted data packet. The other is the packet rate reduction by putting multiple audio frames or mixed audio-video frames in a single AVP packet. As a result, the end-to-end media unit delay is reduced while achieving implicit synchronization. Furthermore, AVP provides adaptive quality of service (QoS) by the prioritized packetization scheme. Simulation results are presented to verify the advantages of the AVP protocol.     

基于实时传输协议的丢包实时修复

在诸如IP电话和远程会议系统等实时应用中,使用实时传输协议RTP(real-timetransportprotocol)在因特网上传输的数据包不可避免地会丢失,极大地影响了传输服务质量.在RTP上增加丢包修复功能可以解决这个问题.介绍了在RTP上采用FEC(forwarderrorcorrection)修复丢包的方法,基于这个方法设计并实现了一个支持丢包修复功能的RTP库,说明了丢包修复功能的测试方法和结果.     

基于GM(1,1)模型的自适应链路层ARQ控制策略

提出了一种用于无线实时流媒体传输的自适应链路层ARQ控制策略,用以提高接收方的播放质量。该策略采用跨层设计的方法,基于GM（1,1）模型预测当前的网络状态,考虑GOP可解码帧数的特性,自适应地调整ARQ参数Nmax;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。     

实时视频通信中的自适应前向纠错方案设计

本文针对实时视频通信中的网络丢包问题，提出了一种基于Reed Solomon算法的自适应FEC方案。与以往的静态FEC编解码方案不同，该方案引入一种新的基于SIP／RTP的QoS反馈机制，根据丢包率大小在发送端调整FEC冗余度、整体发送速率以及封包大小来保证服务质量；并针对网络突发丢包情况，在对数据包进行FEC编码时采用了交织技术     

远程教学直播系统中媒体同步的研究与实现

对于远程教学直播系统这样包含不同媒体流的分布式多媒体应用而言，媒体同步是一项重要内容。利用RTP（实时传输协议）传输机制中的时间戳和序列号信息，提出同步控制算法，实现了流内同步和流间同步。考虑到分组网络带来的延时抖动，算法可以动态地适应网络延时变化，从而保证了分布式环境中媒体同步的服务质量。     

基于错误传播模型的非均等视频流丢失保护

提出一种适用于丢包网络、面向图像组（GOP）层的非均等视频流丢失保护方案。利用GOP中不同帧之间的非均等显著性,将不同数量前向错误校正包分配到GOP层的不同帧中。采用帧间包交错机制将突发包丢失分散到不同帧上,提高处理突发包丢失时的鲁棒性。仿真结果表明,在不同信道丢失模式下,该方案能提高视频接收质量。     

基于AVS M的3G视频流媒体传输与差错控制*

分析了以RTP包传输视频数据,并以RTCP包进行控制的传输机制。根据3GPP规范和相关协议,设计实现了针对AVSM码流的负载包、重传包和六种不同的RTCP包,包括引入的FB包。建立了一个包传输和控制系统。在立即反馈的基础上提出了一种基于分层重传的差错控制机制。     

基于AVS-M的3G视频流媒体传输与差错控制

分析了以RTP包传输视频数据，并以RTCP包进行控制的传输机制。根据3GPP规范和相关协议，设计实现了针对AVS-M码流的负载包、重传包和六种不同的RTCP包，包括引入的FB包。建立了一个包传输和控制系统。在立即反馈的基础上提出了一种基于分层重传的差错控制机制。     

A client-driven media synchronization mechanism for RTP packet-based video streaming

Media synchronization is used to correctly playback a video stream with its associated audio. To support synchronization between video and audio streams transported over IP networks, an RTP/RTCP protocol suite is usually employed. In conventional server-driven media synchronization, the server needs to periodically transmit an RTCP sender report (SR) packet to provide the client with a UTC time in NTP format corresponding to the RTP timestamp carried by each RTP packet. In this paper, we propose a precise client-driven media synchronization mechanism for an RTP packet-based multimedia streaming service. In the proposed method, the server does not need to send any RTCP SR packets for synchronization. Instead, the client device derives the precise normal play time (NPT) for each video and audio stream from the received RTP packets containing an RTP timestamp. Simulations show that the proposed client-driven synchronization method can provide accurate media synchronization without employing an RTCP SR packet and accordingly reduce the initial synchronization delay, the processing complexity at the client device, the number of required user datagram protocol ports, and the amount of control traffic injected into the network.     

Multipoint-to-point communications for SHE surveillance with QoS and QoE management

The bandwidth-hunger applications of SHE (Smart Home Environment) can take advantage of the multipoint-to-point (MPP) connections to aggregate more bandwidth to gain user-perceived Quality of Experience (QoE) and network Quality of Service (QoS). The receiver-centric transport-layer R²CP (Radial Reception Control Protocol) was proposed to resolve the incapability of the MPP communication in conventional TCP and UDP. However, R²CP has no consideration to discriminate the importance in a packet payload which is critical to QoE and brings an issue for critical data packets that may be dropped in great risk of network congestion. In this paper, we thus present P-R²CP (Prioritized R²CP) to effectively decrease the loss ratio of critical data packets in MPP video streaming while the network is congested. P-R²CP is a cross-layer protocol that considers both the transport-layer issues and the media content’s properties in application-layer. Then, an example on MPP-UVS (MPP ubiquitous video surveillance) is presented as UVS is now a very important Internet application that requires QoS/QoE management to protect lives and assets especially in SHE. Our experiments are conducted on different kinds of surveillance videos over MPP links with different bandwidth and packet loss inserted. The experimental results demonstrate that, as the loss of critical packets is decreased by an order and much less critical data packets are dropped, P-R²CP can highly guard not only QoS but also QoE of SHE surveillance video streaming.     

Concurrent multipath transmission combining forward error correction and path interleaving for video streaming

Video streaming is a popular application on next generation networks (NGNs). However, video streaming over NGNs has many challenges due to the high bit error rates of these networks. Forward error correction (FEC) is often applied to improve the quality of video streaming. However, continuous lost packets decrease the recovery performance of FEC protection in NGNs. To disperse continuous lost packets to different FEC blocks, we propose a concurrent multipath transmission that combines FEC with path interleaving. Our proposed control scheme adaptively adjusts the FEC block length and concurrently sends data interleaved over multiple paths. Experimental results with our approach show improved packet loss and signal to noise ratio performance.     

XOR-based frame loss recovery scheme for video streaming

High coding dependencies among video frames suffer from vulnerability to packet loss, which impacts the playback quality of video streaming. In this paper, according to the characteristics of MPEG4/H.264 encoding methods, we propose a simple and low-complexity XOR-based FEC frame loss recovery scheme. Within an entire Group of Pictures (GOP), the proposed scheme shows the ability to recover simultaneously I-frame loss and one P-frame loss. The high frame loss resilience improves the playback QoS of compressed video streaming. The mathematical analysis reveals that the proposed scheme has 72.7% performance improvement than no frame loss protection in term of full GOP frames successful decoding rate.     

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     

一种实时视频流发送算法

通过分析实时视频传输过程中视频帧问存在一定帧间隔、每帧视频编码时闻不同、编码后数据大小不同等特点,提出一种动态调整RTP（实时传输协议）数据包发送间隔的算法,并进行试验分析。试验结果表明．该算法可以有效降低实时视频传输过程中的丢包率和抖动时间。     

Error-resilient surveillance video transmission based on a practical joint source-channel distortion computing model

In recent years, IP (Internet Protocol)-based video surveillance has widely been useful for post-event analysis and assisting the work of privacy protection and public safety. To support high-quality IP video surveillance, error-resilience techniques are important for surveillance system design, because video has more stringent requirements than general video transmission for packet loss, latency, and jitter. The optimal FEC (forward error correction) code rate decision is a crucial procedure to determine the optimal source and channel coding rates to minimize the overall picture distortion when transporting video packets over packet loss channels. The conventional FEC code rate decision schemes using an analytical source-coding distortion model and a channel-induced distortion model are usually complex and typically employ the process of model parameter training, which involves potentially high computational complexity and implementation cost. To avoid the complex modeling procedure, we propose a simple but accurate joint source-channel distortion model to estimate the channel-loss threshold set for optimal FEC code rate decision. Since the proposed model is expressed as a simple closed form and has a small number of scene-dependent model parameters, a video sender of the surveillance system using the model can be easily implemented. For training the scene-dependent model parameters in real time, we propose a practical test-run procedure. This method accelerates the test-run while maintaining its accuracy for training the scene-dependent model parameters. Using the proposed simple model and practical test-run method, the video sender can find the optimal code rate for on-the-fly joint source-channel coding whenever there is a change in the packet-loss condition in the channel. Simulations show that the proposed method can accurately estimate the channel loss threshold set, resulting in an optimal FEC code rate with low computational complexity.     

一种实时视频流发送算法

通过分析实时视频传输过程中视频帧间存在一定帧间隔、每帧视频编码时间不同、编码后数据大小不同等特点,提出一种动态调整RTP(实时传输协议)数据包发送间隔的算法,并进行试验分析。试验结果表明,该算法可以有效降低实时视频传输过程中的丢包率和抖动时间。     

Concealment of whole-frame losses for wireless low bit-rate video based on multiframe optical flow estimation

In low bit-rate packet-based video communications, video frames may have very small size, so that each frame fills the payload of a single network packet; thus, packet losses correspond to whole-frame losses, to which the existing error concealment algorithms are badly suited and generally not applicable. In this paper, we deal with the problem of concealment of whole frame-losses, and propose a novel technique which is capable of handling this very critical case. The proposed technique presents other two major innovations with respect to the state-of-the-art: i) it is based on optical flow estimation applied to error concealment and ii) it performs multiframe estimation, thus optimally exploiting the multiple reference frame buffer featured by the most modern video coders such as H.263+ and H.264. If data partitioning is employed, by e.g., sending headers, motion vectors, and coding modes in prioritized packets as can be done in the DiffServ network model, the algorithm is capable of exploiting the motion vectors to improve the error concealment results. The algorithm has been embedded in the H.264 test model software, and tested under both independent and correlated packet loss models with parameters typical of the wireless environment. Results show that the proposed algorithm significantly outperforms other techniques by several dBs in peak signal-to-noise ratio (PSNR), provides good visual quality, and has a rather low complexity, which makes it possible to perform real-time operation with reasonable computational resources.