Classification based mode decisions for video over network

A video encoder has to make many mode decisions in order to achieve the goal of a low bit rate, high quality, and fast implementation. We propose a general classification based approach to making such mode decisions accurately and efficiently. We first illustrate the approach using the Intra-Inter coding mode decision. We then focus on the decision to skip or code a frame for rate control of video over networks. Using the classification used approach we show improvement in the rate-distortion sense. We then extend the work to scalable video coding in choosing between scalability modes and examine the performance of our approach over error prone networks, using simulated packet losses     

对具有帧间相关性的分层视频码流的非均匀错误保护

在分层视频码流的网络实时传输中,非均匀错误保护（UEP）技术已经被广泛地研究。至今为止,几乎所有的UEP相关工作都是针对具有链式结构的单一码流。但是,在一些实际的图像及视频编码系统（例如MPEG-4）产生的码流中,都存在着更为复杂的树型相关性结构。针对这种具有树型相关性结构的码流提出了一种打包方案,并在此基础上提供了两种带宽分配方案。试验中,采用MPEG-4FGS作为信源编码器。试验结果表明,采用灵活的帧间自适应带宽分配方案可以有效地提高系统性能。     

精细分级视频码流传输和容错性能的研究

精细分级编码是解决网络异构性的有效方法之一。编码输出码流的容错性能是衡量编码技术性能的重要指标。该文实现了一个支持精细分级编码的H.264编解码器,并对精细分级码流在IP网上的传输性能进行了研究。提出了一种精细分级码流容错传输打包策略。实验结果表明,该方案具有良好的容错性能。     

可伸缩多媒体传输错误保护算法综述

可伸缩多媒体对传输中的丢包和误码极为敏感,需要有效的错误保护．主要讨论基于前向纠错的错误保护算法,其核心是利用率失真优化和信源信道联合编码等技术在信源和信道之间合理分配码率,以使接收方重构信号质量最好．依据信道的错误统计特性,将传输信道分为丢包信道和无线误码信道两种,对每种信道模型的错误保护模式和优化目标都给出了统一的形式化问题描述,对主流算法的技术特点和性能指标等进行了分析和比较,最后讨论了下一步研究方向．     

信源-信道联合编码的FGS增强层视频传输研究

研究了基于IP无线网络中精细粒度可伸缩性（FGS）视频的侍输。基于包交换的IP无线网络通常由两段链路组成：有线链路和无线链路。为了处理这种混合网络中不同类型数据包的丢失情况,对FGS视频增强层数据运用了一个具有比特平面间不平等差错保护（BPUEP）的多乘积码前向纠错（MPFEC）方案进行信道编码。对FGS增强层每一个比特平面（BP）,在传输层,采用里德——索罗蒙码（RS）提供比特平面间的保护;而在链路层,则运用循环冗余校验码（CRC）串联率兼容穿孔卷积码（RCPC）提供数据包内保护。还提出了一个率失真优化的信源——信道联合编码的码率配置方案,仿真结果显示出该方案在提高接收端视频质量方面的优势。     

Cross-Layer prioritized H.264 video packetization and error protection over noisy channels

Video transmission over wireless channels is affected by channel-induced packet losses. Distortion due to channel errors can be alleviated by applying forward error correction. Aggregating H.264/AVC slices to form video packets with sizes adapted to their importance can also improve transmission reliability. Larger packets are more likely to be in error but smaller packets require more overhead. We present a cross-layer dynamic programming (DP) approach to minimize the expected received video distortion by jointly addressing the priority-adaptive packet formation at the application layer and rate compatible punctured convolutional (RCPC) code rate allocation at the physical layer for prioritized slices of each group of pictures (GOP). Some low priority slices are also discarded to improve protection to more important slices and meet the channel bit-rate limitations. We propose two schemes. Our first scheme carries out joint optimization for all slices of a GOP at a time. The second scheme extends our cross-layer DP-based approach to slices of each frame by predicting the expected channel bit budget per frame for live streaming. The prediction uses a generalized linear model developed over the cumulative mean squared error per frame, channel SNR, and normalized compressed frame bit budget. The parameters are determined over a video dataset that spans high, medium and low motion complexity. The predicted frame bit budget is used to derive the packet sizes and corresponding RCPC code rates for live transmission using our DP-based approach. Simulation results show that both proposed schemes significantly improve the received video quality over contemporary error protection schemes.     

基于IP无线网络FGS视频传输的多乘积码方案

研究基于IP无线网络中精细粒度可伸缩性(FGS)视频的传输。基于包交换的IP无线网络通常由两段链路组成:有线链路和无线链路。为了处理这种混合网络中不同类型数据包的丢失情况,对FGS视频增强层数据运用了一个具有比特平面间不平等差错保护(BPUEP)的多乘积码前向纠错(MPFEC)方案进行信道编码。对FGS增强层每一个比特平面(BP),在传输层,采用里德—索罗蒙码(RS)提供比特平面间的保护;而在链路层,则运用循环冗余校验码(CRC)串联率兼容穿孔卷积码(RCPC)提供数据包内保护。还提出了一个率失真优化的信源—信道联合编码的码率配置方案,仿真结果显示出该方案在提高接收端视频质量方面的优势。     

Concealment of Whole-Picture Loss in Hierarchical B-Picture Scalable Video Coding

H.264/AVC scalable video coding (H.264/AVC SVC), as the scalable extension of H.264/AVC, offers the flexible adaptivity in terms of spatial, temporal and SNR scalabilities for the generated bitstream. However, such compressed video still suffers from the bad playback quality when packet loss occurs over unreliable networks. In this paper, we present an error concealment algorithm to tackle the whole-picture loss problem in H.264/AVC SVC when hierarchical B-picture coding is used to support temporal scalability. In the proposed algorithm, by taking advantage of the temporal relationship among the adjacent video pictures, the motion information of the lost picture is derived simply and efficiently based on the principle of temporal direct mode. Utilizing the derived motion information, the lost picture is concealed by performing motion compensation on the correctly received temporally previous and future video pictures. The experimental results demonstrate that as a post-processing tool, the proposed error concealment algorithm is able to significantly improve both the objective and subjective qualities of the decoded video pictures in the presence of packet losses when compared to the error concealment algorithm used in H.264/AVC SVC reference software. The proposed method can also be applied to H.264/AVC with hierarchical B-picture coding for error concealment.     

Real-time error concealing bitstream adaptation methods for SVC in IPTV systems

The current heterogeneity in networks and devices demands for a high degree of flexibility in IPTV systems for digital television. A scalable video coding scheme (in this paper we focus on H.264/AVC’s scalable video coding extension SVC) accommodates this flexibility from the coding point of view. Because the IP-based network delivery chain in IPTV systems may suffer from packet loss (having a severe impact on the visual quality) it is necessary to provide means for error concealment. In this paper we propose a novel method that performs adaptation on impaired SVC bitstreams so that the resulting adapted bitstream is compliant to the SVC specification and that the reconstruction result at the decoder is equivalent compared to the approach where the error concealment is implemented in the decoder itself. The adapted bitstreams have a significantly higher visual quality while our approach does not require any modification to existing SVC-compliant decoders. The results of several experiments show that the proposed method is extremely fast (over 900 frames/s) and that it introduces a negligible overhead in terms of bit rate (ca. 0.02%).     

基于心理声学模型的多码率零树小波音频压缩方法

ＭＰＥＧ－４音频编码标准不仅对码率和音质提出了更高的要求,而且还要求编码器具有多种功能以满足各种不同应用的需要,该文利用不同尺度小波系数的自相似特性和人耳的掩蔽效应,提出了一种基于心理声学模型的零树小波音频编码算法。该算法不仅可在低码率（５６ｋｂ／ｓ）上得到透明质量的ＣＤ音频信号,而且可产生嵌入式码流,在最优意义上支持多码率的可分级编码,是一种很有前途的适用一多媒体通信等领域的编码方案。     

基于RCPC-FEC的3G视频传输模型研究

为了有效克服3G无线网络传输中的比特错误和数据包丢失问题,采用码率兼容删除卷积码（RCPC）抑制比特错误问题,通过前向纠错（FEC）减小数据丢包率,将RCPC与FEC有机结合提出了一种基于RCPC-FEC的无线网络视频传输模型。在3G无线网络传输的不同比特错误率和数据丢包率下对该模型进行了测试,实验结果证明了模型的有效性和可行性。     

Transmission distortion estimation for real-time video delivery over hybrid channels with bit errors and packet erasures

We study transmission distortion of video delivery over hybrid channels with bit errors and packet erasures. Hybrid channels have been prone to suffering from transmission error in the forms of packet erasures due to congestion at wired networks, as well as bit errors caused by the wireless interference or fading on wireless links. We present a recursion approach to transmission distortion estimation over hybrid channels in successive frames with the prediction mode. This approach is feasible for video transmission applications as it is capable of online estimation of transmission distortion on received decoded video. Our extensive simulation results on six well-known video sequences demonstrate that our proposed approach is accurate and robust.     

基于SP帧无线视频传输容错编码新方法

视频业务误码敏感度高并且有实时性要求,这与无线信道相对较窄的带宽和较高误码率形成了尖锐的矛盾.因此,要在无线信道上开展视频业务,必须在保证视频编码压缩效率的同时,提高视频编码的容错性能.在对新一代视频编码国际标准H.264中SP/SI帧技术研究的基础上,提出一种SP帧与I帧相结合的抗误码扩散方案CSWI.实验证明,CSWI方法抗误码扩散性能比单一地使用SP帧取代I帧更好.     

Seamless switching of scalable video bitstreams for efficient streaming

Efficient adaptation to channel bandwidth is broadly required for effective streaming video over the Internet. To address this requirement, a novel seamless switching scheme among scalable video bitstreams is proposed in this paper. It can significantly improve the performance of video streaming over a broad range of bit rates by fully taking advantage of both the high coding efficiency of nonscalable bitstreams and the flexibility of scalable bitstreams, where small channel bandwidth fluctuations are accommodated by the scalability of a single scalable bitstream, whereas large channel bandwidth fluctuations are tolerated by flexible switching between different scalable bitstreams. Two main techniques for switching between video bitstreams are proposed. Firstly, a novel coding scheme is proposed to enable drift-free switching at any frame from the current scalable bitstream to one operated at lower rates without sending any overhead bits. Secondly, a switching-frame coding scheme is proposed to greatly reduce the number of extra bits needed for switching from the current scalable bitstream to one operated at higher rates. Compared with existing approaches, such as switching between nonscalable bitstreams and streaming with a single scalable bitstream, our experimental results clearly show that the proposed scheme brings higher efficiency and more flexibility in video streaming.     

一种基于信源信道联合的最优速率分配算法

针对视频数据在无线信道上可靠传输问题,提出了一种基于信源信道联合的最优速率分配算法。该算法在网络带宽一定的情况下,从信源、信道及差错弹性能力权衡考虑,引入了信源解码器的抗误码性能指标,根据不同的信道状态确定信源信道编码的最优速率分配方案,从而获得最大的可解码长度,并最终获取最佳重建视频质量。仿真结果表明,该方案与传统的联合信源信道速率分配算法相比可获得更高的性能增益,适合于视频数据在无线网络上传输。     

无线网络中一种智能控制的组播拥塞控制机制

针对现有组播拥塞控制算法应用到无线网络中存在的性能下降问题,提出一种基于新的智能组播拥塞控制机制ECMCC。ECMCC机制根据网络相对队列时延和数据包丢失检测网络的拥塞状态,采用代表集合机制反馈信息,利用专家控制器的推理判断区分丢包原因和当前的网络状态,进而采取不同的控制策略调节组播源端发送速率。仿真结果表明,ECMCC机制收敛速度快、灵敏性好、速率变化平滑,在有线网络中具有良好的TCP友好性。同时,ECMCC能有效区分网络拥塞和随机差错,提高了网络的吞吐量,适用于无线网络环境,且在无线网络较低误码率时具有一定的TCP友好性。     

无线移动网络中增强TCP性能的技术综述

TCP是因特网上的主要传输协议，在数据包丢失主要是由拥塞引起的传统网络上，TCP可以充分发挥其性能，但是在无线移动网络中，TCP将无线信道比特差错和移动切换引起的数据包丢失误归于网络发生拥塞而采取拥塞控制措施，不必要地降低了端到的吞吐率，导致自身性能的下降，首先阐述了当前无线移动网络的发展状况和TCP面临的问题，在此基础上，综述和评价了各种增强TCP性能的技术方案，最后，结合无线移动通信系统的最新进展，分析和探讨了当前存在的问题和进一步的研究方向。     

改进的MPEG-4静态纹理BQ模式编码方法

从零树符号的信息含义及其与编码层目标图像的一致性出发，研究MPEG-4VTC工具中BQ模式编码(即PEZW方法)的改进问题．提出基于符号分解的：PEZW改进方法——SP-PEZW方法．提出零树符号的分解表示方法，将零树符号中包含的两部分信息用两个分解符号分别表示，并通过删除编码层中与目标图像“不一致”的分解符号来提高编码层的压缩比．实验表明，SP-PEZW方法与PEZW方法相比，低分辨率编码层有更高的压缩比，低分辨率空间层以指定码率解码的图像质量有明显提高；高分辨率编码层的压缩比并没有明显下降，而高分辨率空间层以指定码率解码的图像质量略有提高．更重要的是，SP-PEZW方法的实现只是在PEZW方法中增加一个零树符号分解编码环节，同时还继承了PEZW方法的随机读取、有容错性、可实现局部编码、内存需求小等多种重要特性。     

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.     

Digital video watermarking based on intra prediction modes for audio video coding standard

Digital video watermarking provides means for carrying information targeted for synchronization, error resilience or copyright protection. However, it is difficult to get a good trade-off between the embedding capacity, imperceptibility and efficiency. In this paper, a novel digital video watermarking algorithm based on intra prediction modes of AVS (audio video coding standard) is proposed. The algorithm hides one bit in each qualified intra 8 × 8 luma block by modifying intra 8 × 8 prediction modes based on the mapping rules between the modes and hidden bits. The specific positions of the host 8 × 8 blocks are determined by the features of every block and a position template indicated by the key. Watermark information can be retrieved by decoding the intra prediction modes from bitstream, requiring neither original media nor complete video decoding. Experimental results show that the proposed algorithm has little impact on video quality and video stream features. A comparatively high embedding rate is obtained with little impact on bit rate.