Multicast scheduling for VOD services

Providing quality-of-service- (QoS-) guaranteed video on demand (VOD) services over wireless access in vehicular environments (WAVEs) is a challenge as WAVE adopts enhanced distributive channel access (EDCA), a contention-based channel access mechanism, for air interface access control. This paper proposes a selective downlink scheduling (SDS) algorithm to enhance the quality of VOD for WAVE networks. According to the importance of video decoding, video packets are categorized into high and low priorities. The categorized packets are put into different queues in roadside units (RSUs) to contend for transmission opportunities. Aiming to improve video playback quality and reduce video playback delay, the proposed SDS algorithm schedules video packets based on their importance, playback deadline, and their real-time parameters of receiving onboard units (OBUs), such as velocity and remaining dwelling time. The effectiveness of SDS algorithm is verified by simulations.

The integration of computers and communications is able to automate storage, processing and transportation of information. Video information is considered most bulky as it needs high space and consequently high bandwidth for real time playback. Thereby, Video on Demand (VOD) systems are inherently resource intensive and demanding with respect to the performance for continuous playback. The data stream is to be maintained between server and consumer for longer periods once playback of video begins. The restricted bandwidth of all contemporary media and the need to maintain long duration connectivity have been major bottleneck for jitter and breaks free VOD services. In this paper, an analysis and evaluation of buffer requirements for jitter and break free playback of video from a streaming server has been presented. In practice, the ordered delivery of data packets through public domain networks is not guaranteed, which can break the sequential playback of streamed video data. The performance of VOD system thus is critically dependent upon buffer available with the user. The limits and bounds for continuous playback versus buffer and real transmission plan have been illustrated, which have been identified by the authors during their recent research work cited under references. This paper also overviews the video streaming and playback processes for VOD systems and provides for graphical and formal representation of limits and bounds for buffers.

In recent years, there has been a strong interest in transmitting compressed video over packet switched networks, such as asynchronous transfer mode (ATM). Previous work has dealt with transmitting MPEG over constant bit rate (CBR) and variable bit rate (VBR) services of ATM. The available bit rate (ABR) service of ATM is expected to be much more cost effective than CBR or VBR. However, there hasn't been much work done on running interactive client/server applications (for example, video on demand) over ABR. We have developed a framework to design interactive video systems transmitting MPEG video over the ATM ABR service. We have developed models to determine the network connection parameters required to run interactive client/server multimedia applications over an ATM network using the ABR service. We solve our model using real-time dynamic equilation (RTDE) analysis. We conclude that by proper dimensioning of the buffers at the client and the server, it is possible to run interactive video over the ATM ABR service.

In Peer-to-Peer (P2P) Video-on-Demand (VoD) streaming systems, supporting free VCR operations is challenging. Prefetching is a good way to improve user experience of VCR interactivity. But most existing P2P VoD prefetching schemes are proposed aimed at those popular videos with large amount of log data, without considering the situation that the videos are unpopular or the popular videos are in their initial phase of release. In this situation, these schemes cannot support user VCR interactivity very well. To address this issue, we propose a new optimized prefetching scheme, called Hybrid Anchor Scheme (HAS), in which fixed anchors and dynamic anchors are merged together. The dynamic anchors are generated based on association rule and segments popularity. Through combining the way of sequential prefetching according to weight of segments and the way of several rounds prefetching, we implement HAS effectively. Extensive simulations validate the proposed prefetching scheme provide shorter seeking latency compared to other prefetching schemes.

A store-and-forward architecture is presented that can provide video-on-demand (VOD) as well as other database distribution services. It assumes a B-ISDN network to be in place. The four major elements in this architecture are the information warehouse (IWH), where video material is archived; the central office (CO) server, which contains a processor responsible for service management and a video buffer that interacts directly with network customers; and the customer premise equipment. A requested video program is provided in a real-time fashion from the CO server to the customer. At the information warehouse the video program is retrieved from the archival storage in blocks, and with transfer rates much faster than real-time. Subsequently, it is sent in a bursty mode to the CO servers via high speed trunks. >

This survey is an analysis about different Video-On-Demand protocols. They have been classified depending on the mode of service that they have. After analysing them, the most popular two developed systems have been explained: Youtube and CNN Pipeline. Video-On-Demand is a service that enables users to select a video (or a movie) from a large collection while sitting at home and permits to have almost as a good a control over the viewing of the video as when using a conventional VCR.

Advances in fiber optics, video compression and transmission technologies are driving the development and deployment of a new generation of CATV equipment. Optical fiber super- trunks are now expanding program delivery capabilities to over 150 channels and digital video compression will soon increase the capacity of existing satellite, fiber and cable systems by a factor of 5 to 10 or more. These next generation CATV systems will support Video on Demand (VOD) and advanced communications services such as broad band LAN/WAN networking and Personal Communications Services (PCS).

A synchronization problem, which always arises when sound, video, motion pictures and other media are brought together and integrated into a computer system, is one of the most important issues in multimedia communications and applications. In a time-sharing and multi-process environment such as the UNIX operating system, the traditional synchronization mechanism results in two fatal defects, namely, audio discontinuity and out of synchronization between the audio and video signals. A novel media synchronization model in a multi-process environment is proposed. Based on this model, a continuous media playback module was implemented and served as the key component of the two multimedia systems developed in the Communication and Multimedia Laboratory of National Taiwan University. One is the multimedia authoring system which provides interactive VCR-like operations and audio/video editing facilities. The other is a prototype of the VOD (video on demand) system which provides video browsing facility. Both systems show that the performance of the proposed synchronization model is quite satisfactory. >

The growing popularity of high resolution video and the continuously increasing demands for high quality video on mobile devices are producing stronger needs for more efficient video encoder. Concerning these desires, HEVC, a newest video coding standard, has been developed by a joint team formed by ISO/IEO MPEG and ITU/T VCEG. Its design goal is to achieve a 50% compression gain over its predecessor H.264 with an equal or even higher perceptual video quality. Motion Estimation (ME) being as one of the most critical module in video coding contributes almost 50%-70% of computational complexity in the video encoder. This high consumption of the computational resources puts a limit on the performance of encoders, especially for full HD or ultra HD videos, in terms of coding speed, bit-rate and video quality. Thus the major part of this work concentrates on the computational complexity reduction and improvement of timing performance of motion estimation algorithms for HEVC standard. First, a new strategy to calculate the SAD (Sum of Absolute Difference) for motion estimation is designed based on the statistics on property of pixel data of video sequences. This statistics demonstrates the size relationship between the sum of two sets of pixels has a determined connection with the distribution of the size relationship between individual pixels from the two sets. Taking the advantage of this observation, only a small proportion of pixels is necessary to be involved in the SAD calculation. Simulations show that the amount of computations required in the full search algorithm is reduced by about 58% on average and up to 70% in the best case. Secondly, from the scope of parallelization an enhanced TZ search for HEVC is proposed using novel schemes of multiple MVPs (motion vector predictor) and shared MVP. Specifically, resorting to multiple MVPs the initial search process is performed in parallel at multiple search centers, and the ME processing engine for PUs within one CU are parallelized based on the MVP sharing scheme on CU (coding unit) level. Moreover, the SAD module for ME engine is also parallelly implemented for PU size of 32×32. Experiments indicate it achieves an appreciable improvement on the throughput and coding efficiency of the HEVC video encoder. In addition, the other part of this thesis is contributed to the VLSI architecture design for finding the first W maximum/minimum values targeting towards high speed and low hardware cost. The architecture based on the novel bit-wise AND scheme has only half of the area of the best reference solution and its critical path delay is comparable with other implementations. While the FPCG (full parallel comparison grid) architecture, which utilizes the optimized comparator-based structure, achieves 3.6 times faster on average on the speed and even 5.2 times faster at best comparing with the reference architectures. Finally the architecture using the partial sorting strategy reaches a good balance on the timing performance and area, which has a slightly lower or comparable speed with FPCG architecture and a acceptable hardware cost

Owing to the traffic characteristics of a video-on-demand (VOD) system, it is very attractive to use the isochronous service for VOD traffic transmission. This paper proposes a mechanism which uses an isochronous service to support a video-on-demand service over WDM networks. Based on the proposed mechanism, the problem of scheduling isochronous and asynchronous traffic with a variable number of receivers and transmitters for single-star WDM networks is investigated. For only asynchronous traffic, an optimal scheduling algorithm is presented; for both isochronous and asynchronous traffic, a heuristic algorithm is proposed for the single-star WDM networks with multiple receivers and transmitters. Simulation results show that the average switching duration and the average number of switching modes obtained by the proposed algorithm are efficient.

There is a renewed interest in video-based communications. The expected proliferation of various video communication applications has in turn resulted in a need for scalable and efficient digital video encoding schemes which would enable the dissemination of video signals at differing spatial resolution, depending on end-user needs and limitations. Also, along with this is the need for a universal broadband in which the same transmitted signal is efficiently distributed to various end-user applications, allowing them to communicate at differing grades of video quality. As a first step, this paper presents an efficient and scalable video coding scheme. The scheme is capable of encoding and decoding video signals in a hierarchical, multilayer fashion to provide video at differing quality grades, e.g., NTSC, extended quality (EQTV), high definition (HDTV). Since the lower levels in such a scheme can be encoded independently of the upper levels, compatibility with some of the standard codecs can be directly imposed. In addition, strategies to control quantization noise for independent transmission by means of residual layering are also discussed. This paper also discusses the utilization of this coding scheme to enable efficient bit rate sharing and robust distribution of video communication signals at three distinct quality grades in multicasting, as well as multipoint communication applications for multiparty video-conferencing. Two special distribution architectures are considered which involve the use of SMDS and ATM as the transport and delivery vehicles.

In this contribution, we derive a spatio-temporal extrapolation method for 3-D signals. Lost areas in video signals caused by transmission errors in video communications are concealed by extrapolating the surrounding video signal into the missing area. The method is integrated into the H.264/AVC coder as concealment feature. By exploiting spatial and temporal correlations at the same time, it is possible to inherently compensate motion. Deviations in luminance occurring from frame to frame can be compensated, too. Since the spatio-temporal selective extrapolation method does not rely on motion vectors, the algorithm can be also easily applied to Intra concealment by additionally exploiting temporal correlations. Further, the frequency selective extrapolation shows especially convincing results in connection with flexible macroblock ordering (FMO) causing a chequerboard like loss pattern in case of transmission errors

MPEG-2 video communications over ATM networks is one of the most active research areas in the field of computer communications. In the transmission process of a variable bit rate video signal over an ATM network, cells are inevitably exposed to delays, errors and losses due to the statistical multiplexing used in these networks. These phenomena affect the quality of the video signal and without adequate measures to control the propagation of the impairments the quality of the service may fall below acceptable levels. In this paper, we study the impact of cell losses on the quality of a MPEG-2 video sequence encoded in a variable bit rate mode. We introduce a set of control mechanisms at different levels of the protocol architecture to be used in MPEG-2-based video communications systems using ATM networks as their underlying transmission mechanism. Our results (using different video sequences) show the effectiveness to improve the video quality by using a structured set of control mechanisms to overcome for the loss of cells carrying VBR MPEG-2 video streams. We argue that in order to be able to create video systems able to cope with cell losses encountered in computer communications systems, a structured set of error-resilient protocol mechanisms is needed.

Transmission of MPEG2-encoded video is one of the most demanding applications in terms of network resources and QoS requirement. It needs high bandwidth with stringent transmission delays. It can not tolerate large variations on delays and it requires low error and loss data rates. Therefore, in order to design efficient integrated video communication systems over ATM networks, we propose in this paper to analyze the effects of errors and delays on both video signal and network performance.

The problem of error control and concealment in video communication is becoming increasingly important because of the growing interest in video delivery over unreliable channels such as wireless networks and the Internet. This paper reviews the techniques that have been developed for error control and concealment. These techniques are described in three categories according to the roles that the encoder and decoder play in the underlying approaches. Forward error concealment includes methods that add redundancy at the source end to enhance error resilience of the coded bit streams. Error concealment by postprocessing refers to operations at the decoder to recover the damaged areas based on characteristics of image and video signals. Last, interactive error concealment covers techniques that are dependent on a dialogue between the source and destination. Both current research activities and practice in international standards are covered.