Video communication is often afflicted by various forms of losses, such as packet loss over the Internet. This paper examines the question of whether the packet loss pattern, and in particular the burst length, is important for accurately estimating the expected mean-squared error distortion. Specifically, we (1) verify that the loss pattern does have a significant effect on the resulting distortion, (2) explain why a loss pattern, for example a burst loss, generally produces a larger distortion than an equal number of isolated losses, and (3) propose a model that accurately estimates the expected distortion by explicitly accounting for the loss pattern, inter-frame error propagation, and the correlation between error frames. The accuracy of the proposed model is validated with JVT/H. 26L coded video and previous frame concealment, where for most sequences the total distortion is predicted to within -4-0.25 dB for burst loss of length two packets, as compared to prior models which underestimate the distortion by about 1.5 dB. Furthermore, as the burst length increases, our prediction is within -4-0.7 dB, while prior models degrade and underestimate the distortion by over 3 dB.

This paper addresses the problem of video coding in a joint source-channel setting. In particular, we propose a video encoding algorithm that prevents the indefinite propagation of errors in predictively encoded video---a problem that has received considerable attention over the last decade. This is accomplished by periodically transmitting a small amount of additional information, termed coset information, to the decoder, as opposed to the popular approach of periodic insertion of intra-coded frames. Perhaps surprisingly, the coset information is capable of correcting for errors, without the encoder having a precise knowledge of the lost packets that resulted in the errors. In the context of real-time transmission, the proposed approach entails a minimal loss in performance over conventional encoding in the absence of channel losses, while simultaneously allowing error recovery in the event of channel losses. We demonstrate the efficacy of the proposed approach through experimental evaluation. In particular, the performance of the proposed framework is 3--4 dB superior to the conventional approach of periodic insertion of intra-coded frames, and 1.5--2 dB away from an ideal system, with infinite decoding delay, operating at Shannon capacity.

Despite the well-known challenges of variations in throughput, delay, and packet loss over the Internet, video streaming has experienced phenomenal growth, owing to the extensive research in video coding and transmission. In this paper, we review several recent advances for channel-adaptive video streaming that, we believe, will benefit the design of video streaming systems in the future. Employed in different components of the system, these techniques have the common objective of providing efficient, robust, scalable and low-latency streaming video. First, by allowing the client to control the rate at which it consumes data, adaptive media playout can be used to reduce receiver buffering and therefore average latency, and provide limited rate scalability. Secondly, rate-distortion optimized packet scheduling, a transport technique, provides a flexible framework to determine the best packet to send given the channel behaviors, the packets' deadlines, their transmission histories, the distortion reduction associated with sending each packet, and the inter-packet dependencies. Thirdly, at the source encoder channel-adaptive packet dependency control can greatly improve the error-resilience of streaming video and reduce latency. Finally we address the specific additional challenges for wireless video streaming. We consider three architectures for wireless video and discuss the utility of the reviewed techniques for each architecture.

The use of digital video offers immense opportunities for creators; however, the ability for anyone to make perfect copies and the ease by which those copies can be distributed also facilitate misuse, illegal copying and distribution (“piracy”), plagiarism, and misappropriation. Popular Internet software based on a peer-to-peer architecture has been used to share copyrighted movies, music, software, and other materials. Concerned about the consequences of illegal copying and distribution on a massive scale, content owners are interested in digital rights management (DRM) systems which can protect their rights and preserve the economic value of digital video. A DRM system protects and enforces the rights associated with the use of digital content. Unfortunately, the technical challenges for securing digital content are formidable and previous approaches have not succeeded. We overview the concepts and approaches for video DRM and describe methods for providing security, including the roles of encryption and video watermarking. Current efforts and issues are described in encryption, watermarking, and key management. Lastly, we identify challenges and directions for further investigation in video DRM.

This article examines the challenges that make simultaneous delivery and playback, or streaming, of video difficult, and explores algorithms and systems that enable streaming of pre-encoded or live video over packet networks such as the Internet

When designing a system for video communication over a lossy packet network, it is highly beneficial to have a mechanism for accurately predicting the mean-squared error (MSE) distortion that results from different packet loss patterns. This paper proposes Distortion Chains model for accurately predicting the end-to-end distortion for different general packet loss patterns. The performance is examined using JVT/H.264 encoded video sequences and previous frame error concealment. It is shown that for all tested sequences the proposed model predicts the total distortion due to a packet loss pattern within a 10 % error bound 80 % of the time, as compared to the conventional additive approach which achieves the same accuracy less then 40 % of the time.

Abstract—Layered multicast is an efficient technique to deliver video to heterogeneous receivers over wired and wireless networks. In this paper, we consider such a multicast system in which the server adapts the bandwidth and forward-error correction code (FEC) of each layer so as to maximize the overall video quality, given the heterogeneous client characteristics in terms of their end-to-end bandwidth, packet drop rate over the wired network, and bit-error rate in the wireless hop. In terms of FECs, we also study the value of a gateway which “transcodes ” packet-level FECs to byte-level FECs before forwarding packets from the wired network to the wireless clients. We present an analysis of the system, propose an efficient algorithm on FEC allocation for the base layer, and formulate a dynamic program with a fast and accurate approximation for the joint bandwidth and FEC allocation of the enhancement layers. Our results show that a transcoding gateway performs only slightly better than the nontranscoding one in terms of end-to-end loss rate, and our allocation is effective in terms of FEC parity and bandwidth served to each user. Index Terms—Layered video multicast, optimal bandwidth allocation, optimal FEC, transcoding and nontranscoding gateways, wireless Internet. I.

Abstract—We present a technique for low-complexity ratedistortion (R-D) optimized adaptive video streaming based on the concept of rate-distortion hint track (RDHT). RDHTs store the precomputed characteristics of a compressed media source that are crucial for high performance online streaming but difficult to compute in real time. This enables low-complexity adaptation to variations in transport conditions such as available data rate or packet loss. An RDHT-based streaming system has three components: 1) information that summarizes the R-D attributes of the media; 2) an algorithm for using the RDHT to predict the distortion for a feasible packet schedule; and 3) a method for determining the best packet schedule to adapt the streaming to the communication channel. A family of distortion models, denoted distortion chains, are presented which accurately predict the distortion produced by arbitrary packet loss patterns. Two distortion chain models are examined which lead to two RDHT-based techniques. We evaluate the proposed techniques for two canonical problems in streaming media, adaptation to available data rate and to packet loss. Experimental results demonstrate that for the difficult case of nonscalably coded H.264 video, the proposed systems provide significant performance gains over conventional low-complexity streaming systems, and achieve this gain with a comparable level of complexity making them suitable for online R-D optimized streaming. Index Terms—Distortion modeling, hint track (HT), low-complexity, multimedia streaming, rate-distortion (R-D), video adaptation, video coding. I.

Internet transmission is characterized by variations in throughput, delay, and packet loss, which can severely affect the quality of multimedia presentations delivered over the network. Still, Internet video streaming has experienced phenomenal growth in the last few years, owing to the extensive research in video coding and transmission. In this paper, we review several recent advances for network-adaptive video streaming that, we believe, will benefit the design of video streaming systems in the future. Employed in different system components, these techniques have the common objective of providing efficient, robust, scalable and low-latency streaming video. They range from purely server or source encoder-based techniques, through transmission schemes that could be implemented either at the sender or at the receiver, to purely client-based techniques. We discuss each of them in detail, presenting also related work and experimental results. We end the paper with a summary of the reviewed techniques and a brief discussion of future research directions.

Video simulcasting enables a sender to generate replicated streams of different rates, serving receivers of diverse access bandwidths. As replication introduces noticeable redundancy, balancing bandwidth consumption with user satisfaction becomes a critical concern in simulcasting. This paper investigates the above issue; more explicitly, we seek answers to the following two questions: what is the number of streams that should be generated, and what is the bandwidth that should be allocated to each stream? We derive optimal and efficient solutions, and evaluate their performance under a variety of configurations. The results demonstrate that an optimal and adaptive bandwidth allocation significantly improves user satisfaction under stringent resource constraints, and an optimal choice of the stream number yields further improvements.