素材迁移转码技术在备播系统中的实现

为了保障电视播出的安全,许多电视台投入了备播系统,但也面临着新旧不同品牌视频服务器并存的境况。本文介绍了素材转码迁移的基本原理,并对不同类型的视频服务器之间转码迁移过程进行研究,并以SeaChange服务器中迁移视音频素材文件到Harris服务器为例解构迁移转码系统。     

数码视讯推出运营级N屏合一集群转码系统

近期,作为中国数字电视及三网融合龙头企业,数码视讯推出了N屏合一集群转码解决方案,这是一个运营级的视音频转码解决方案。该系统囊括了视听新媒体领域主流视音频编码格式、数据封装格式,不但支持直播点播类的流媒体转码,而且具备高性能文件并发转码能力,在低码率和保证优秀视频输出质量的条件下全面支持F2F/F2S/S2S/S2F自由互     

浅析IP电视中的视频转码技术

IP电视也就是网络电视,这是一种向家庭用户提供包括数字电视在内的多种交换式服务的新技术,IP电视的出现有效的帮助了不同宽带使用用户通过互联网观看数字电视节目。IP电视中的视频转码技术的实施,有效的避免了用户宽带的限制,并且提高了转码的效率。本文对IP电视和视频转码技术分别进行了全面的分析,并探讨了IP电视中视频转码方案,进一步分析了IP电视中的视频转码技术。     

一种基于视频转码的IP电视应用

本文介绍了一种将卫星数字电视通过网络进行直播的IP电视应用方案,通过采用像素域级联的视频转码结 构以及运动重估计技术,确保了视频转码的质量,并大大提高了视频转码的效率。     

浅谈FFmbc及其在广电行业中的应用

在电视节目制作中,经常需要将各种各样的流媒体文件转换为广电行业使用的格式,使用非编软件进行转码是常用的一种方法,而本文介绍另一种更便捷实用的方法,即通过转码软件FFmbc,将流媒体文件转码为广播级视频服务器支持的视音频格式.经测试,转码产生的mxf格式文件,在Mediadeck视频服务器上成功播出.     

数码视讯推出N屏合一集群转码系统

近期,数码视讯推出运营级N屏合一集群转码解决方案,系统功能丰富,囊括了视听新媒体领域主流视音频编码格式、数据封装格式,不但支持直播点播类的流媒体转码,而且具备高性能文件并发转码,在低码率和保证优秀视频输出质量的条件下全面支持F2F／F2S／S2S／S2F自由互转。     

索贝中标SMG超高清融合媒体交换平台

近日,索贝成功中标SMG超高清融合媒体交换平台。该平台将利用索贝分布式视频处理框架,兼容不同格式媒体文件,实现对现有文件分布式转码以及不同格式、分辨率参数的转换,结合索贝和华为联合开发的智能分布式对象存储系统,实现超高清素材的高速转码、闪拼及分发迁移。4K媒体内容交换平台将提供分片转码功能,支持HDR/SDR互转、支持超分辨率转码应用,拥有良好的可扩展性。同时4K媒体内容交换平台转码设备将能够全面支持主流专业视音频格式。     

数字电视转码技术的应用

时代在进步,社会在发展。数字电视转码技术也跟上了时代发展的步伐,MPEG更是成为了压缩视频的标准。为了满足用户的不同需求,往往要进行相应的转码。电视转码是压缩编码领域的重要组成成分。     

DCT域视频转码技术综述

随着数字视频技术的广泛应用,实现各种视频编码格式之间转码的要求越来越迫.视频转码是数字媒体网络网关的关键技术.本文从DCT域视频转码的体系结构、关键技术及最优转码策略三个方面,介绍了现有的各种算法,分析和总结了各自的特点,提出了DCT域视频下采样帧内刷新体系结构和最优视频转码策略解的概念,并指出了下一步的研究方向.     

方正旌旗非编系统中文件的输入及输出

文件的输入、输出在非线性编辑系统中占据重要位置。如何将各种不同格式的视音频文件输入到非编系统中,供节目制作的需要,又如何将编辑好的节目打包合成为各种不同格式的视音频文件,以供各种途径的需要,所有的非编产品有不同的实现手段也各有特色。在方正旌旗非线性编辑系统中,对不同格式文件的输入,提供了相应的格式转换程序转码工具。和方正旌旗系统编辑格式不同的各类视音频文件,都可以通过转码工具转换过来。从菜单“工具”中选择“转码工具”,弹出转码窗口(如图1所示)。方正旌旗中的转码工具是完全开放式的程序设计,只要用户安装了相应…     

分布式信源编码错误漂移消除机制的研究

传统视频编码标准由于具有较高的视频压缩效率而得到广泛使用,但是该方法存在错误漂移的现象,而分布式信源编码是当前解决这一现象行之有效的方法。目前提出的错误漂移消除机制采用固定间隔的方法,即每隔几帧选取一帧用于阻止错误漂移,但是该方法在选取视频帧的时候没有考虑被选取视频帧与前后帧的相关性,缺乏灵活性,针对这一问题提出了基于相关性的错误漂移消除选取机制。实验结果证明,该方法可以获得较好的解码效果。     

非线性编辑系统中视频采集丢帧问题探讨

在非线性编辑系统中，视频采集丢帧现象是个常见的问题。从非线性编辑系统的软硬件设置及视频源等方面进行较详细的分析，并提出相应的解决方法，使视频采集的丢帧现象减到最小。     

Robust scalable video multi-cast with multiple sources and inter-source network decoding in lossy networks

This paper presents a robust scalable video multi-cast scheme with source diversity and inter-source network decoding in lossy networks. The source diversity technique gives path diversity, providing a better quality of layered video transmission under hostile environments. For each source, an optimization formulation is set up to find the best transmission route of each transmitting video layer. The objectives of the formulation are to maximize the total information values of video layers reflecting the end-to-end video quality and transmission reliability. The source providing the best overall achievable data rate, which is the data rate destination can expect to receive from the transmission, is selected to be the primary source, while the rest will be secondary sources. When the Quality-of-Service (QoS) guarantees of some transmitting video layers cannot be fulfilled by the primary source, the secondary source with the best QoS parameters is selected to transmit the layers to destinations. The number of secondary sources used for transmissions is increased until the QoS guarantees of all transmitting video layers are satisfied or all network resources are utilized. Network coding is deployed to multi-cast video layers from the same source for efficient resource usage. Network coded data from different sources can be used to decode the transmitting video data. In other words, at each destination, it needs only a sufficient number of video packets from different sources to recover all transmitting video data. Simulations with different network topologies show the improvement in both objective and subjective qualities of layered video multi-cast under lossy environments.     

The transmission of MPEG‐2 VBR video under usage parameter control

This paper studies the transmission of MPEG‐2 VBR video over ATM network under usage parameter control. The idea is to seek a compromise between the network utilization and the quality of video service by applying UPC‐based rate control strategies to the video source. A modified leaky bucket algorithm is proposed to calculate the constraints on the bit‐rate guaranteeing conformance to peak cell rate, sustainable cell rate and burst tolerance usage parameters. Two rate control strategies, one for real‐time generated video coding and the other for pre‐compressed video, are proposed for MPEG‐2 VBR video. The rate control strategies control the video source to generate traffic conforming to the constraints on the bit rate. The experimental results show that both the UPC‐based rate control strategies can provide lossless transmission from the source perspective as well as to reduce the burstiness of the traffic. To keep within the bit‐rate allowed, the control method uses coarser quantization to maintain better picture quality than that by removing the number of AC transformed coefficients. The slight degradation of picture quality caused by the source rate control is preferable than the severe drop of picture quality caused by the cell loss at UPC. Copyright © 2001 John Wiley & Sons, Ltd.     

Streaming video over wireless channels: Exploiting reduced-reference quality estimation at the user-side

We propose a source rate control scheme for streaming video sequences over wireless channels by resorting on a reduced-reference (RR) quality estimation approach. It works as follows: the server extracts important features of the original video, which are coded and sent through the channel along with the video sequence and then exploited at the decoder to compute the actual quality; the observed quality is analyzed to obtain information on the impact of the source rate at the given system configuration; at the receiver, decisions are taken on the optimal source rate to be applied next at the encoder to maximize the quality as perceived at the user-side. The rate is adjusted on a per-window basis to compensate low-throughput periods with high-throughput periods so as to avoid abrupt video quality changes, which can be caused by sudden variations in the channel throughput. The use of the RR quality estimation represents the main novelty of the proposed work. This has the advantage of allowing the rate control to optimize the user-perceived video quality after all the streaming system impairments have affected the signal, including actual channel errors, playback buffer starvation occurrences and error concealment. This approach is new in this context, since in the past proposals video models are used to predict the relationships of the quality with the coding rate, channel errors and starvation occurrences. Numerical simulations show how the proposed approach is able to achieve results similar to those obtained with model-based approaches, but with the significant benefit of not requiring any knowledge on the signal and channel characteristics.     

SAVE: an algorithm for smoothed adaptive video over explicit ratenetworks

Supporting compressed video efficiently on networks is a challenge because of its burstiness. Although a large number of applications using compressed video allow adaptive rates, it is also important to preserve quality as much as possible. We propose a smoothing and rate adaptation algorithm for compressed video, called SAVE, that is used in conjunction, with explicit rate based control in the network. SAVE smooths the demand from the source to the network, thus helping achieve good multiplexing gains. SAVE maintains the quality of the video and ensures that the delay at the source buffer does not exceed a bound. We show that SAVE is effective by demonstrating its performance across 28 different traces (entertainment and teleconferencing videos) that use different compression algorithms     

基于高阶深层时空信息的自媒体视频质量评价

自媒体视频中普遍存在的失真扭曲现象,给视频质量评价带来了新的挑战。视觉感知理论研究表明,人类视觉系统中存在感知迭代机制,即视频的感知评价是一个正反向迭代修正的过程。受此启发,本文将该机制引入视频质量评价,提出了基于高阶深层时空信息的质量评价方法。具体而言,本文提出了二阶协方差聚合来进行高阶空域信息的提取,引入快速迭代GRU结构进行深层时域信息建模,而后通过特征层的池化聚合以及多层感知机回归视频得分。实验结果表明,预测结果与人类主观质量评分具有较好的一致性,明显优于现有无参考质量评价算法。      

IP网络信源信道联合编码策略

面向IP网络,讨论了一种信源编码的失真估计模型,并用典型的视频编码器作了验证.提出了一种基于包的信道编码方法,并将该方法引入信道编码的失真估计中,对一种信道编码的失真估计模型作了改进.以前述内容为基础,提出了两种信源信道联合编码策略,实验结果表明,这两种策略得到的最优结果在很大程度上提高了视频重建质量.     

Quality control for VBR video over ATM networks

Uncontrolled variable-bit-rate (VBR) coded video yields consistent picture quality, but the traffic stream is very bursty. When sent over ATM networks, cell losses may be incurred due to limited buffer capacity at the switches; this could cause severe picture quality degradation. Source rate control can be implemented to generate a controlled VBR bit stream which conforms to specified bit rate bounds and buffer constraints. However, source rate control could result in picture quality degradation too. Hence, for real-time video services, an important issue to address is whether the picture quality degradation incurred by source rate control is within acceptable levels or how to choose the appropriate coding parameters to make it so. We establish quantitatively the relationship between picture quality and source rate control for the case of guaranteed service with different combinations of allocated bandwidth, buffer size, and other key video-coding parameters of MPEG-2. In addition, quality control in the context of two-layered scalable video service (basic and enhanced quality) is also considered. Our study reveals that, in order to maximize both the basic and the enhanced quality, source rate control should be implemented on both layers. The relationships between the two types of quality and different combinations of allocated bandwidths, buffer sizes, and some key coding parameters are also established quantitatively for MPEG-2 SNR scalability     

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.