Real-time video watermarking techniques robust against re-encoding

This paper presents a real-time watermarking codec that is robust against re-encoding attacks for high-definition videos. The codec uses a segmentation function and texture detector techniques for applying real-time watermarking to human visual systems. Experimental results confirm that the proposed scheme satisfies the requirements of invisibility, real-time processing, and robustness against format conversion and low bit-rate encoding. The proposed algorithm has the advantages of simplicity, flexibility, and low computational burden; thus, it is a suitable candidate for many novel and interesting applications such as video fingerprinting for set-top boxes, Internet protocol television, personal video recorders, and satellite boxes.     

Content-based scalable H.263 video coding for road traffic monitoring

For sending video data through very low bit-rate mobile channels, video codec with high compression rate is the pre-requisite. Although the H.263 video codec is recommended as one of the candidates due to its simplicity and efficiency, it is generally believed that its compression efficiency can be further improved if the content-based scalable video coding technique can be applied. In this paper, we propose a modified H.263 encoder which supports real-time content-based scalable video coding. The proposed technique is applied to real-time video surveillance systems for road traffic monitoring. For the proposed approach, the moving objects, i.e. cars, are first extracted from the steady background. Their activities are then further classified as fast or slow by assessing the regularity of their motion. The information is then passed to a modified H.263 encoder to reduce the temporal and spatial redundancies in the video. As compared with the conventional H.263 encoder using for the same application, the proposed system has a 20% increase in compression rate with negligible visual distortion. The proposed system fully complies with the ITU H.263 standard hence the encoded bit stream is completely comprehensible to the conventional H.263 decoder.     

基于RTP/RTCP协议的IP视频系统设计与实现

RTP／RTCP是网络多媒体应用中的核心协议之一，ITU－T在制定分组网上的多媒体通信标准H．323时将RTP／RTCP作为保证多媒体流QoS的关键协议。主要讨论互联网上实时视频通信的设计和实现，在H.323终端中视频采用H.263编码，编码后的码流采用RTP／RTCP协议来打包传送。试验结果表明，该方案的图像连续性较好，传输延迟小，主观效果比较理想。     

基于H.264的改进阈值的可变自适应十字搜索算法

H.264视频编解码标准的算法复杂度高,不能满足视频监控和无线传输中要求的高压缩比、低码率和实时性的要求,通过分析可知运算量最大的是运动估计模块,本文对该模块的算法进行优化。在ARPS算法的基础上结合预测的初始点和自适应调节臂长的搜索策略的优点,采用可变阈值的早期终止策略和简单的搜索模板的算法。实验表明该算法在有效减少搜索点数的情况下并不会对图像质量产生较大的影响,提高了运动估计算法的搜索速率和鲁棒性。     

基于MPEG-4的视频流传输框架

视频的实时传输存在带宽、传输延迟、包丢失率等方面的要求,而当前的Internet环境下未提供任何QoS(服务质量)服务,所以一个有效的视频传输系统必须自适应于网络状况的动态变化。在充分发掘MPEG-4特性的基础上,该文构造了一个基于MPEG-4的视频流传输框架。通过使用带宽预测、自适应速率调整、SFEC纠错、错误隐藏等机制使视频的传输能够自适应于网络状况的动态变化。同时,提出了一种基于优先级的速率调整算法,克服了传统基于R-D理论的速率调整算法对媒体源支持单一的缺点。试验结果表明,该系统在低比特率、网络状况动态变化的环境中能取得良好的视频传输效果。     

无人机多跳自组网实时视频传输性能评估

基于当前无人机广泛采用的M-JPEG与MPEG-2视频编解码器,构建了实时半实物仿真环境,并基于此对无人机多跳自组网实时视频传输进行了性能评估。结果表明,实时条件下的M-JPEG编解码器无法支持多跳实时视频传输,而MPEG-2能够支持2跳至3跳的实时视频传输;在1~10跳的实时传输条件下,MPEG-2较之M-JPEG在分组投送成功率和解码率意义下的视频质量平均分别提高了55.69%和205.64%。相对于传统的单飞单控方式,无人机多跳自组网在将无人机系统的作用范围扩大4~9倍的同时能够支持实时视频传输。     

面向移动网络的实时视频转码系统

针对宽带IPTV视频源向移动网络和移动终端提供视频服务需要解决的视频转码问题,设计开发以软件方式实现的实时视频转码系统,可将MPEG-4格式的高码率、高分辨率节目源实时转换为适合移动网络和移动终端的低码率、低分辨率视频。在误差补偿转码框架的基础上,提出一种自适应的转码时延保证方法,解决了转码计算复杂性和转码质量均衡的关键技术问题,保证了转码系统的实时性。测试结果表明,实现的转码系统视频转换质量损失少、实时性高。     

基于流媒体Location Free电视的设计及实现

本文提出了一种基于流媒体Location Free电视的解决方案,既可以实现一般的电视收看功能,而且直接通过内置的视频网络传输模块连接在异地的电视机,收看在异地的电视频道。在解决方案中,视频采用了目前压缩率最高的H.264编解码算法,音频采用目前使用最广的G.711的编解码算法,并使用网络的流媒体传输协议,进行实时处理,从而实现异地电视信号实时传输通信。     

Parallel video streaming optimizing network throughput

In the Internet, video streaming services, in which users can enjoy videos at home, are becoming popular. Video streaming with high definition TV (HDTV) or ultra high definition video (UHDV) quality will be also provided and widely demanded in the future. However, the transmission bit-rate of high-quality video streaming is quite large, so generated traffic flows will cause link congestion. In the Internet, routes that packets take are determined using static link weights, so the network productivity, i.e., the maximum achievable throughout by the network, is determined by the capacity of a bottleneck link with the maximum utilization, although utilizations of many links remain low level. Therefore, when providing streaming services of rich content, i.e., videos with HDTV or UHDV quality, it is important to flatten the link utilization, i.e., reduce the maximum link utilization. We propose that ISPs use multiple servers to deliver rich content to balance the link utilization and propose server allocation and server selection methods for parallel delivery. We evaluate the effect of parallel delivery using 23 actual commercial ISP networks.     

一种适用于DVB Over IP的传输流传送方法

对DVB Over IP应用来说，如何传送用可变比特率（VBR）压缩的MPEG-2传输流格式的视频是一个重要的问题。其中以传统的恒定速率（CBR）方式传送VBR视频时，接收端需要较大的缓存和开始延时时间；而另一种新的采用PCR协助的恒定速率（PCBR）传送方式，则是以较高的传输速率和以牺牲码率的稳定性来减少缓存需求。为了平衡传输速率和接收端缓存需求，在PCBR基础上，提出了一种称为IPCBR的改进的视频流传送方法。通过实验证明，该新方法与传统的CBR传送方式相比，不仅可降低缓存需求，而且与PCBR方式相比，又具有较高的稳定性和带宽利用率，并克服了PCBR丢包率较高的问题。同时由于该算法特别针对DVB Over IP，具有较强的实用性。且有利于Internet TV业务拓展。     

基于自动镜头分割的视频压缩编码算法

将镜头分割引入压缩编码,针对分割镜头特性确定适当的编码器及参数,能够提高视频压缩编码的效率．为此,提出了一种基于对分查找的镜头分割算法(VSSB)以及关键帧频率计算公式．其算法的核心是通过对分查找形成二叉树,同时通过广度优先搜索寻找分割点,具有O(log n)的时间复杂性．针对实时应用,采用了双阈值比较的镜头分割算法,能够较好地实现镜头切变和渐变的检测．基于镜头分割算法,建立并实现了并行方式的编解码系统模型．实验证明,基于镜头分割的编码方法能够提高压缩比,缩短压缩编码时间,有利于改善重建图像质量。     

基于ARM11的H.264硬编解码视频传输系统设计

为了解决视频信息冗余、视频传输系统成本较高,视频编解码效率低等缺陷,研究了一种基于H.264硬编解码器的视频传输系统;该系统采用以ARM11为核心的,包含多媒体硬编解码器MFC的S3C6410作为处理器,使用CMOS摄像头采集实时视频数据;在嵌入式Linux操作系统上构建基于live555的RTSP服务器,监听客户端请求;重点阐述了多媒体硬编解码器MFC对视频数据进行的H.264硬件压缩编码,以及数据包的RTP封装,然后经由live555流媒体服务器转发至PC机;最后,在PC上对接收到的数据流进行解码播放,经实验测试证明,系统设计稳定可靠,具有可扩展性强、性能高、成本和功耗低等优点,图像质量和时延满足实际应用需求。     

基于多链路聚合的无线实时视频传输系统

无线实时视频传输已成为当前通信领域发展的热点,但当前的无线通信技术提供的带宽尚无法满足高码率的实时视频传输的需求.针对这一问题,设计并开发一个基于多链路聚合技术的无线实时视频传输系统.该系统通过聚合多链路带宽,对各链路运用TFRC速率控制,有效地提高数据传输速率和稳定性,同时采用提早丢弃和关键帧优先发送等实时性优化策略,获得更好的视频传输质量.     

基于WebRTC的实时视音频通信研究综述

实时网页通信(WebRTC)无需在网页上安装插件或客户端即可实现实时视音频通信,具有跨平台、开发成本低和适用范围广等优点,已被W3C纳为草案,正在带来多媒体通信的一场革命.首先介绍基于WebRTC实时视音频通信的总体架构;接着详细阐述其相关关键技术,包括信令机制、网页应用和浏览器底层WebRTC,后者还包括视、音频编译码模块和传输模块;然后从两用户和多用户的角度对WebRTC实时视音频通信应用的实现细节进行比较;最后讨论值得进一步研究的开放问题.     

The design and simulated performance of a mobile video telephonyapplication for satellite third-generation wireless systems

The design and performance of a low bit-rate video telephony service for mobile third-generation (3G) systems is presented. The ITU-T G.723.1 speech coding and the ITU-T H.263 video coding recommendations are used, as proposed by the ITU-T H.324 low bit-rate multimedia communications recommendation. The target bit-rate for the H.324 service is 64 kb/s. The design is performed in conjunction with that of a wideband-code division multiple access (W-CDMA) radio transmission technology (RTT) system, proposed by the European Space Agency (ESA) for the satellite component of the ITU IMT-2000 standard. Most of the results could also be applied to the 3G terrestrial systems. The use of concatenated channel coding with convolutional inner coding and Reed-Solomon (RS) outer coding is investigated. Service designs based on equal error protection (EEP) and unequal error protection (UEP) schemes for the audio and video sources are compared. The simulation of the proposed video telephony services shows that significantly more graceful video and audio degradation is obtained with the proposed UEP scheme than with a more straightforward EEP method. The UEP scheme reduces significantly the occurrence of highly annoying audio and video artefacts, allowing satellite-based video telephony services that are compatible with the current Internet-based applications     

Intelligent copyright protection system using a matching video retrieval algorithm

Many video service sites headed by YouTube know what content requires copyright protection. However, they lack a copyright protection system that automatically distinguishes whether uploaded videos contain legal or illegal content. Existing protection techniques use content-based retrieval methods that compare the features of video. However, if the video encoding has changed in resolution, bit-rate or codec, these techniques do not perform well. Thus, this paper proposes a novel video matching algorithm even if the type of encoding has changed. We also suggest an intelligent copyright protection system using the proposed algorithm. This can serve to automatically prevent the uploading of illegal content. The proposed method has represented the accuracy of 97% with searching algorithm in video-matching experiments and 98.62% with automation algorithm in copyright-protection experiments. Therefore, this system could form a core technology that identifies illegal content and automatically excludes access to illegal content by many video service sites.     

视频实时加密算法的研究与实现

提出一种新的视频实时加密算法RMVEA。RMVEA建立在数字多媒体实时动态系统上,独立于任何视频压缩算法,为实时数字视频提供高强度、高速度的加密。实验结果表明该锋法在基本不增加视频编码器负载及系统延时的情况下,能有效解决视频压缩率与密钥选择自由度的矛盾。     

数据驱动的并行视频编码器的设计和实现(英文)

通用视频编码系统一般基于标准视频压缩算法, 比如 MPEG4 和 H.264。但这些标准算法包括如离散余弦变换和运动估计等高复杂度的计算,所以为了实现实时通信,编码系统通常采用以硬件为基础的实现方法。然而以硬件为基础的视频编码系统具有实施周期长、成本高、灵活性差等不足。文章介绍了一种基于低计算复杂度自适应块截短编码方案的数据驱动并行视频编码系统,实验结果验证了编码系统的有效性。     

Design and implementation of a teleconferencing system using improved HEVC coding

Teleconferencing has become an indispensable element in any business system, because it offers the opportunity for collaborators to participate in a virtual group while remaining in divergent regions. Teleconferencing also increases productivity, minimizes travel expenses and saves travel time. This paper presents a reliable Teleconference system that utilizes an improved high-efficiency video codec (HEVC) H.265 technology with congestion control. The improvement is based on CU size decision and entropy coding, which provides an adequate approach to enhance the real-time video/IP technology in terms of improved video quality and increased compression ratio compared to the previous codec (H.264) and original HEVC.     

Entropy coders and 3D-Hadamard coefficients sequency scan order for a fast embedded color video codec

This work compares the performances of two Golomb family entropy coders applied to a video codec system named FHVC (fast hadamard video codec). The entropy coders considered have different operation modes and specific adaptation strategies. The work also presents a new 3D-transform coefficient scan order developed for the FHVC. This scan process is based on the multiplication of the three-dimensional sequency numbers of each coefficient. The FHVC (which is also described in this work) is a fast embedded color video codec developed to be implemented in a video set-top box used in a fiber optics network. The focus is on more reduced execution times, and not on higher compression rates. Low computational complexity and use of meager computational resources are also required. All the multiplications and divisions operations are performed by binary shifts and the system is implemented exclusively with 16-bit integer arithmetic. Even with these constraints, good distortion versus bit-rate results were achieved. The Hadamard transform is used in a three-dimensional fashion, in order to reduce spatial and temporal correlation and to avoid costly motion estimation and compensation techniques. The proposed scan procedure allows the transform coefficient reading in an idealistic “decreasing in the average” order. After the scan procedure, the encoding of the bit sequence of the 3D-Hadamard coefficients is carried out, bit-plane-by-bit-plane, with an adaptive Golomb run-length entropy coder, which produces a fully embedded output bitstream. Two entropy coders were considered. The first one uses an empirical, but fast and efficient, adaptation strategy that shows good results on non-stationary data. The second one has an adaptation strategy that is nearly optimum, in a maximum-likelihood sense, for independent Bernoulli identically distributed data.