一种基于H.263的实时软件编解码器实现方案

利用VC6技术设计了一套集图像的实时采集、实时编码和实时解码为一体的H．263软件编解码演示系统，在编码算法的具体实现中增加了零块判断并结合人的视觉特性改进了码率的分配方案。从实验结果看，系统能够实时采集符合H.263标准的各种分辨率格式视频序列，且对其中一些格式可以实时编码，是一种良好的软件编解码器实现方案。     

基于DM642的X264视频监控系统的软件实现

视频监控系统对视频压缩率和编码实时性具有很高的要求，X264算法的高压缩率和DM642处理器的高性能正好满足这两个需求。文中分析了基于DM642的X264视频监控系统的软件实现，讨论了基于RF5的算法移植过程和移植后的程序优化，并给出了QCIF格式的实时视频编码的优化结果。     

窄带可视电话设备中的视频编解码器

根据ＩＴＵ的Ｈ．２６１建议，建立了视频编解码器的模型，并给出了视频压缩编码算法的具体参数。在软件模拟的基础上得出了一些有益的结论，理论分析与实验结果相符合。     

基于IP的MPEG-4视频编码器设计及其在应急通信中应用

介绍了MPEG-4视频压缩标准及视频采集、编码原理，以数字信号处理芯片DSP TMS320C6211构建平台．设计了应用于矿山救援应急多媒体通信中，基于IP传输的MPEG-4视频编码器硬软件，重点讨论算法优化方法．并给出实际应用结果。     

基于H.264的猪舍实时视频监控系统的研究

设计了一种基于H.264的猪舍实时视频监控系统。为了提高视频传输的实时性,对目前H.264编码算法中性能优越的UMHexagons算法进行了改进,主要针对十字形搜索、5×5正方形搜索和六边形搜素模板进行改进。实验结果表明,改进后的算法在不影响图像质量的前提下,降低了视频编码的复杂度,减少了视频编码的时间,使得实时性得到了显著的提高。     

多接口视频编码方案的设计与实现

针对现有编码系统处理视频格式单一的问题,提出多接口视频编码方案,采用TI DSP(TMS320DM8168)+FPGA构架,将多种常见的视频接口设计在同一系统中,应用FPGA采集视频数据后传送给DM8168,通过软件控制送至各子模块,进行编码。经长期测试结果表明,该系统编码性良好,视频显示无误码,适用性和通用性更强。     

动态环境下视频兴趣区的自动捕获系统研究

传统的动态环境下视频兴趣区自动捕获和监控识别系统采用Zig Bee编码和特征提取方法,因动态视频在传输过程中产生衰减失真,导致视频图像采集传输丢包,自动捕获性能不好。提出一种基于Huffman编码和MUX101程控开关控制的动态环境下视频兴趣区的自动捕获系统。通过VXI系统总线高速传输视频数据到AD8021芯片,进行反馈电阻控制,实现视频特征的动态提取和捕获,选用程控放大器VCA810,通过局部总线向HP E1562E 8GB提供传感器信号,达到调整视频特征兴趣区域放大倍数的目的。设计Huffman编码及视频兴趣区特征提取算法作为软件核心部分嵌入,采用VXI总线技术实现对动态视频兴趣区的自动捕获系统的硬件设计和软件设计。仿真结果表明,采用该系统进行动态视频采集和兴趣区特征的自动捕获,准确度较高,丢包率较少,性能优越。     

基于DM6467的视频监控系统设计

本文实现了一个基于DM6467的标清和高清的视频监控系统,包括标清/高清模拟视频输入输出、H.264标清/高清视频编码、AAC音频编解码、H.264标清/高清视频实时解码和基于TCP协议的网络传输等功能,其不仅满足不同情况下的视频监控需求,还具有良好的可重构性和可扩展性。本文主要讨论了系统硬件和软件的设计。     

基于IP多播技术的网络视频监控系统的研究

围绕IP多播技术在网络视频监控系统中的应用，深入讨论了系统中引入的IP多播技术所涉及到的一系列关键技术。根据网络视频监控系统原理图，使用UML语言构架出系统软件平台，给出实现基于IP多播技术的网络视频监控的一般算法，通过对WinSock套接字的实现，完成视频监控系统的网络通信软件的设计。     

视频传输中的码率控制技术

码率控制是视频通信系统中的一个天键环节，主要用于调摔编码视频的输出质量。设计码率摔制系统时一般需要涉及目标码率计算、信源与信道模型建立、视频编码方法、率失真估计、码率分配、跳帧、实时性、传输环境和系统性能评估等诸多方面。新出现的网络视频流媒体传输、无线信道传输、MPEG-4的对象编码传输、信噪比精细可分级编码传输等实际应用要求鲁棒高效的码率控制算法来满足视频实时传输的需要。相应的码率摔制算法已成为近几年的研究热点。本文给出了视频码率控制技术的系统框架和基本要求，介绍了现有的各种恒定码率(CBR)与可变码率(VBR)摔制算法，综合比较了各自的优缺点，概括论述了视频码率控制技术的发展现状，并给出了下一步的研究方向。     

Application of H.263+ Video Coding Modes in Lossy Packet Network Environments

The quality of real-time audio and video information transmitted via today's Internet suffers severely from often significant packet losses. While this problem is well understood and solved for existing audio coding schemes, support from the video coding standards themselves is required for video streams. This paper presents the newly introduced error resilience mechanisms built into the second version of H.263 (1998), known under its working name H.263+, and addresses the corresponding packetization format issues that together significantly improve the image quality at packet loss rates up to 20%. In particular, it is support from the video coding algorithm itself, paired with appropriate transport layer mechanisms, that leads to significant improvements of perceived image quality for communicative as well as retrieval applications at moderate bit rates up to some 100 kbit/s.     

A robust coding scheme for packet video

Some of the important characteristics and requirements of packet video are discussed. A layered packet video coding algorithm based on a progressive transmission scheme is presented. The algorithm provides good compression and can handle significant packet loss with graceful degradation in the reconstruction sequence. A network simulator used in testing the scheme is introduced, and simulation results for various conditions are presented     

Bit-rate control algorithm on video packet coding and its characteristics

A bit-rate control method based on a hybrid coding algorithm is proposed for packet video coding. Three types of bit-rate control modes, the constant bit-rate mode, the average bit-rate control mode and the free bit-rate mode, are investigated. According to the opinion test the latter two modes have an advantage in video quality over the constant bit-rate mode. A teleconference terminal, which includes this video codec, audio codec and packet adaptor, is implemented in actual hardware on High-speed Packet Switching system. On this system, video information are transmitted in UI frame (a packet format in X.25 protocol) at 64 K 800 Kbit/s.     

Layered coding for ATM based video distribution systems

This paper discusses packet loss and its protection in an asynchronous transfer mode (ATM) based video distribution system. Packet losses in ATM based networks have such a great impact on the design of coding algorithms and network architectures that they should be exhaustively discussed and resolved. In this paper, first basic configuration of the ATM based video transmission system and its packet-loss protection schemes are discussed. The DCT based layered coding scheme with packet priority classification is proposed as an effective packet-loss protection scheme. Burstiness characteristics of the broadcast video sources are evaluated and modeled to clarify statistical multiplexing performance and packet-loss properties. The quality degradation caused by the packet losses is also evaluated by the SNR, and the superior performance of the proposed layered coding scheme is verified.     

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.     

H.264编码环中的去块效应滤波系统

介绍了H.264的编解码模型系统中的去块效应滤波系统，分析了该系统原理及其相对于以往去块效应滤波系统的改进。并通过仿真实验验证了该系统在提高图像质量和降低视频流码率上的较好作用。     

Distributed and Adaptive Analog Coding for Video Broadcast in Wireless Cooperative System

Recently, some analog joint source-channel coding (AJSCC) schemes have been proposed to deal with cliff effect in wireless video broadcasting system. And wireless video broadcast with user cooperation tends to be a promising way to improve broadcast video quality in the near future. In this paper, we introduce a distributed and adaptive analog coding scheme called ACVC (adaptive cooperative video coding) based on AJSCC and with the concept of coset coding in distributed source coding, to improve the overall video broadcast quality in wireless cooperative system. Particularly, an adaptive packet discarding module is introduced to the framework to avoid video quality deterioration under severe channel conditions. And a model for quantization step selection of coset coding is built to minimize the redundancy in the cooperative signal and improve the anti-noise ability of the video. The experimental results show that, ACVC has stronger adaptability and thus obtains higher quality of broadcasted video than existing wireless cooperative schemes in the literature under different channel conditions.     

Variable bit-rate coding of video signals for ATM networks

Statistical characteristics of video signals for video packet coding, are clarified and a variable-bit-rate coding method for asynchronous transfer mode (ATM) networks is described that is capable of compensating for packet loss. ATM capabilities are shown to be greatly affected by delay, delay jitter, and packet loss probability. Packet loss has the greatest influence on picture quality. Packets may be lost either due to random bit error in a cell header or to network control when traffic is congested. A layered coding technique using discrete-cosine transform (DCT) coding is presented which is suitable for packet loss compensation. The influence of packet loss on picture quality is discussed, and decoded pictures with packet loss are shown. The proposed algorithm was verified by computer simulations     

高清视频联网系统中一种新的丢包率分析方法

为了提高系统性能并最大限度降低算法实现复杂度,本文提出了一种新的高清视频联网系统中丢包率的更优近似分析方法,重点研究包长度对丢包率的影响。首先使用马尔可夫链模型估算数据包的误码率,进而得出数据包网络时延的概率分布,最后计算出基于包长度的丢包率的理论表达式。仿真结果表明,通过减小发包大小可以有效降低丢包率,同时基于包长度的丢包率的理论分析方法可以有效地运用到高清视频联网系统的分析研究中。