Video Communication

Receiver adaptation layered multicast video has been pro- posed to address the issue of network bandwidth heterogeneity in multi- party video communication over networks. An important issue in this ap- proach is how to conduct layer adaptation to achieve the "optimal level" of subscription to maximize the perceived video quality at receivers while uti- lize bandwidth efficiently. In the current work, optimal layers are typically determined by congestion signals fed back from networks. This approach implies that instantaneous congestion is a key parameter; however, in ran- dom traffic, such an implication somewhat under-estimates other parame- ters. In order to investigate the performance of "layer adaptation" in random traffic for improving the network-based multi-party video communication system, in this paper we present an analytical studying of the issue of re- ceiver layer adaptation for layered video transmission. In this work, we use the notion of congestio...

In this paper, we present an intelligent multi-hop communications architecture for quality video delivery over networks. The key to this architecture is the insertion of relays with different levels of intelligence along the end-to-end path that coordinate with each other and the end-systems. The main objective is to achieve a synergy among these strategically deployed relays and end-systems to optimize the end-to-end performance. Toward that end, we study traffic management and content adaptation within this architecture. Compared with the end-to-end model, the multi-hop model can achieve higher throughput without increasing the risk of overloading the network and the end-to-end delay. Preliminary experimental results show a throughput gain of 19.3% on a concatenation of two best-effort networks, and 37.8% for a concatenation of a best-effort network and a guaranteed service network. The content adaptation translates this throughput gain into 0.9dB gain in PSNR on average. With the error withdrawal feature at the relay, abrupt quality drops are avoided.

Virtual communities have a potential to improve the community building process and care services through support for activities, participation and information needs. In this paper, we propose a mobile virtual community platform for elderly healthcare. The requirements for this platform are collected from interview sessions with clients and caregivers in a care institution, and a workshop with multiple stakeholders. We focus on the technical platform that fulfills the requirements gathered from this approach. The platform is an elaboration of our existing mobile virtual community platform and enables tailoring through an adaptable set of mobile and platform services. Reminder services, remote monitoring, video communication and (delegated) friend and service finding are discussed and exemplified through a case study.

We present a system integrating gesture and live video to support collaboration on physical tasks. The architecture combines network IP cameras, desktop PCs, and tablet PCs to allow a remote helper to draw on a video feed of a workspace as he/she provides task instructions. A gesture recognition component enables the system both to normalize freehand drawings to facilitate communication with remote partners and to use pen-based input as a camera control device. Results of a preliminary user study suggest that our gesture over video communication system enhances task performance over traditional video-only systems. Implications for the design of multimodal systems to support collaborative physical tasks are also discussed.

We describe our system called MobileASL for real-time video communication on the current U.S. mobile phone network. The goal of MobileASL is to enable Deaf people to communicate with Sign Language over mobile phones by compressing and transmitting sign language video in real-time on an off-the-shelf mobile phone, which has a weak processor, uses limited bandwidth, and has little battery capacity. We develop several H.264-compliant algorithms to save system resources while maintaining ASL intelligibility by focusing on the important segments of the video. We employ a dynamic skin-based region-of-interest (ROI) that encodes the skin at higher quality at the expense of the rest of the video. We also automatically recognize periods of signing versus not signing and raise and lower the frame rate accordingly, a technique we call variable frame rate (VFR). We show that our variable frame rate technique results in a 47% gain in battery life on the phone, corresponding to an extra 68 minutes of talk time. We also evaluate our system in a user study. Participants fluent in ASL engage in unconstrained conversations over mobile phones in a laboratory setting. We find that the ROI increases intelligibility and decreases guessing. VFR increases the need for signs to be repeated and the number of conversational breakdowns, but does not affect the users' perception of adopting the technology. These results show that our sign language sensitive algorithms can save considerable resources without sacrificing intelligibility.

Novel bitwise retransmission schemes are devised which retransmit only the bits received with small reliability. The retransmissions are used to accumulate the reliabilities of individual bits. Unlike the conventional automatic repeat request (ARQ) schemes, the proposed scheme does not require a checksum for the error detection. The bits to be retransmitted are reported as a combination number, or two synchronized random number generators (RNGs) at the transmitter and receiver are used to greatly compress the feedback message. The bitwise retransmission decisions and/or combining can be performed after the demodulation or after the channel decoding at the receiver. The bit-error rate (BER) expressions are derived for the case of one and two retransmissions, and verified by computer simulations. Assuming three specific retransmission strategies, the scheme parameters are optimized to minimize the overall BER. For the same number of retransmissions and packet length, the proposed schemes always outperform the frequently used stop-and-wait ARQ. The impact of feedback errors is also considered. Finally, practical designs of the bitwise retransmissions for data fusion from sensor nodes in Zigbee, Wifi and Bluetooth networks are presented.

Novel bitwise retransmission schemes are devised which retransmit only the bits received with small reliability. The retransmissions are used to accumulate the reliabilities of individual bits. Unlike the conventional automatic repeat request (ARQ) schemes, the proposed scheme does not require a checksum for the error detection. The bits to be retransmitted are reported as a combination number, or two synchronized random number generators (RNGs) at the transmitter and receiver are used to greatly compress the feedback message. The bitwise retransmission decisions and/or combining can be performed after the demodulation or after the channel decoding at the receiver. The bit-error rate (BER) expressions are derived for the case of one and two retransmissions, and verified by computer simulations. Assuming three specific retransmission strategies, the scheme parameters are optimized to minimize the overall BER. For the same number of retransmissions and packet length, the proposed sche...

Mobile phones have become essential part of modern life and continue to change the way people communicate with each other. The camera phones have become ubiquitous in the recent past and video communication services such as videophone are getting ...

Alhamdulillah, thanks God, without H~m I am nothing. I would like to express my sillcere thanks to my major adviser Dr. Mitchell L. Neilsen for his guidanceẽ ncouragement, help, and support for the completion of my thesis. Without his constant support and inspiring ideas, this thesis would have been impossible. I also would like to thank Dr. K.M. George and Dr. H. Lu for serving on my graduate committee and for their precious time spending on my thesis. I also want to thank Dr. J.P. Chandler for his generous support a.nd help. I would also like to thank to my Indonesian Government represented by The Agency for The Assessment and Application of Technology (BPP Teknologi) for giving me an opportunity to pursue my education and sponsoring me. To Nusantara Aircraft Industries Limited (PT IPTN) where I have been working for, I want to thank for aU support that has been given to me and for giving me an opportunity to get my dream pursuing to the higher education. Last~but certainly not least, I wish to express with the deepest of my h art my sincere thanks to my parents, the ones who always instil in me the important of learning and always pray to God for me. My thanks also go to all of my friends who have helped, encouraged, and advised me during my stay in Stillwater.

In this paper, a permutation scheme is introduced for Distributed Video Coding. The main goal is to preserve privacy and security in video surveillance video communications and can be adapted for other scenarios. The proposed approach consists to apply permutation based encryption to the DVC scenario. The permutation is defined by a secret key which is required at the decoder for decompression providing security. Simulation results show that the reference architecture and the permutation-based proposed have similar rate distortion performance.

Low prices for high-performance personal computers and fiber networks with increasing amounts of bandwidths have created the opportunity for new multimedia applications. Audio and video streams, telephony and multiplayer real-time computer games, for example, ...

In this paper, we present t-Room, the next generation video communication system we are developing. Our approach is to build rooms with identical layouts, including walls of display panels on which users and physical or virtual objects are all shown at life-size. In this way, the user space enclosed by t-Room's surrounding displays can be shared as a common space at any other site. In other words, the enclosed spaces overlap each other. This configuration effectively provides symmetric reproduction of the audio-visual information surrounding local and remote users and objects. The feeling provided by t-Room is different from that by conventional videoconferencing systems, since there is no spatial barrier separating users such as the video screen of a conventional videoconferencing system. Furthermore, t-Room benefits in every way from Next Generation Network (NGN) technology: QoS, service productivity, and security. We view t-Room as a future form of telephone service.

We propose a 2D display and camera arrangement for video communication systems that supports both spatial information between distant sites and user mobility. The implementation of this arrangement is called the "surrounding back screen method." The method enables users to freely come from and go into other users' spaces and provides every user with the direct pointing capability, since there is no apparent spatial barrier separating users, unlike the case of conventional video communication systems. In this paper, we introduce two properties ("sharedness" and "exclusiveness") and three parameters (a distance and two angles) to represent the geometrical relationship between two users. These properties and parameters are used to classify the shared spaces created by a video communication system and to investigate the surrounding back screen method. Furthermore, to demonstrate and explore our surrounding back screen method, we have developed a prototype system, called t-Room. Taking into account practical situations, we studied a practical case where two t-Rooms with different layouts are connected.

We propose a novel video mediation method that immerses remote users in a virtual shared space. In the implemented system using this method, video cameras and screens surround users, and on the screens placed behind them remote users and physical or virtual ...

Even with the availability of videoconferencing systems (VCSs), there are still occasions when there is no substitute for direct face-to-face discussions. Conventional VCSs have aimed to create a strong feeling of presence by using high-quality audio and video and virtual reality ...

Mobile 3D video communication, especially with 2D and 3D compatible, is a new paradigm for both video communication and 3D video processing. Current techniques face challenges in mobile devices when bundled constraints such as computation resource and compatibility should be considered. In this work, we present a wireless 2D and 3D switchable video communication to handle the previous challenges, and name it as Stereotime. The methods of Zig-Zag fast object segmentation, depth cues detection and merging, and texture-adaptive view generation are used for 3D scene reconstruction. We show the functionalities and compatibilities on 3D mobile devices in WiFi network environment.

This study investigates co-speech movements as a function of the conversational turn exchange type, the type of speech material at a turn exchange, and the interlocutor's role as speaker or listener. A novel interactive protocol that mixes conversation and (non-read) nursery rhymes works to elicit many speech turns and co-speech movements within dyadic speech interaction. To evaluate a large amount of data, we use the density of cospeech movement as a quantitative measure. Results indicate that both turn exchange type and participant role are associated with variation in movement density for head and brow co-speech movement. Brow and head movement becomes denser as speakers approach overlapping speech exchanges, indicating that speakers increase their movement density as an interruptive exchange is approached. Similarly, head movement generally increases after such overlapping exchanges. Lastly, listeners display a higher rate of co-speech movement than speakers, both at speech turns and remote from them. Brow and head movements generally behave similarly across speech material types, conversational roles, and turn exchange types. On the whole, the study demonstrates that the quantitative co-speech movement density measure advanced here is useful in the study of co-speech movement and turn-taking.

Abstract. In a computer supported distant collaboration, communication between users can be enhanced with a visual channel. Plain videos of individual users unfortunately fail to render their joint actions on the objects they share, which limits their mutual perception. Remote interaction can be enhanced by immersing user representations (avatars) and the shared objects in a networked 3D virtual environment, so user actions are rendered by avatar mimicry. Communication gesture (not actions) are captured by real-time computer vision and rendered. We have developed a system based on a single-webcam for body and face 3D motion capture. We have used a library of communication gestures to learn statistical gesture models and used them as prior constraints for monocular motion capture, so improving tracking ambiguous poses and rendering some motion details. We have developed an open source library for real-time image analysis and computer vision that supports acceleration by consumer grap...

Virtual communities have a potential to improve the community building process and care services through support for activities, participation and information needs. In this paper, we propose a mobile virtual community platform for elderly healthcare. The requirements for this platform are collected from interview sessions with clients and caregivers in a care institution, and a workshop with multiple stakeholders. We focus on the technical platform that fulfills the requirements gathered from this approach. The platform is an elaboration of our existing mobile virtual community platform and enables tailoring through an adaptable set of mobile and platform services. Reminder services, remote monitoring, video communication and (delegated) friend and service finding are discussed and exemplified through a case study.

Today’s mobile devices have become increasingly powerful with enhanced features such as increased CPU power and memory, internet connectivity in multiple ways (multi-homing) and interfacing with external peripheral devices (for instance GPS receiver, medical sensors). The proliferation of these mobile devices combined with an increasing willingness of users to share information available on and around mobile device (e.g. location, user

Ayman Abdalla as well for being my local supervisor. I thank him for his consistent support and encouraging comments during the research stages. I also thank Dr. Shakeel Ahmad for the useful discussion on the research and for his advice on dealing with problems. I thank my colleagues Dr. Muneeb Dawood and Mr. Shadan Khattak for their cooperation and help. I want to thank my beloved De Montfort University and all its familiar staff for the interesting time I spent with them. A special thanks to my colleagues and friends for their help and support: Dr. Osama Abdulhay for the statistical analysis and Dr. Mohammad Daoud and Mr. Hani al-Mimi for their help in the subjective evaluation. I also thank all staff and students of the Department of Multimedia Systems at Al-Zaytoonah University. v Dedication I dedicate this work to the soul of my father who supported me during all stages of my life. I dedicate this work to my mother for encouraging and giving me the power to tackle any task during the PhD. I also dedicate the work to my brothers, sisters and fiancé for their patience and prayers. I would like also to dedicate this thesis to my uncles Dr. Abdelfatah

This paper proposes a new idea which makes the processes not to wait for a long time to enter into a critical section to access a data structure which is already assigned to the other process and that process enters into critical section. Till date so many approaches and algorithms are proposed to reduce the delay time between the successive executions of the critical section, but no one proposed how to assign the data structure to all the process who are requesting for the data structure. In this paper the idea we are proposing is to create an instance of the data structure before assigning to the process and allow each and every process into critical section to avoid the delay time permanently.  The  proposed  algorithm  every  time  creates  an instance of the data structure before assigning to the requested process and simultaneously it checks for the instance released by the process already assigned instanced data structure. If when a new process is requesting for the data structure and at the same time already assigned process releases the instance then the coordinator will do both creation and deletion of the instance.

The paper describes the development of a new methodological approach to the study of the features enabling the simulation of face-to-face (FTF) interactions using Advanced Video Conferencing (AVC) technologies. The methodology is based on the notion of Grounded Theory and utilises the study of users of the technology for discovering the criteria needed for the investigation of AVC as simulating FTF. The methodology enabled the development of an informed scheme of criteria which will be used in customising an annotation system suitable for the study of AVC interactions. The paper illustrates initial experimentations with various existing annotation systems in a search for further exploring the needs outlined by the methodology. The paper describes the implementation of the new methodology in a New Zealand based case study which investigated the reasons underpinning uptake of the technology across nationally dispersed research staff and students.

Video streaming demands high data rates and hard de-lay constraints, and it raises several challenges on to-day's packet-based and best-effort internet. In this paper, we propose a simple multiple-description coding (MDC) technique based on sub-sampling and cubic-spline interpolation to provide spatial diversity, such that no additional buffering delay or storage is required. Analysis of the frame dropping rate due to packet loss and drifting error under the multi-path streaming envi-ronment is further explored in this paper.

1. Introduction Intelligent Space or iSpace is a new kind of computing system aiming at improving the environments of humans, creating a natural and easy to use solution. Its main feature is that the intelligence is not implemented separately in the actors, but it is distributed in the whole space. In this paper, we present a hand gesture and movement recognition system, whose purpose is to be used as an intuitive interface for the iSpace.

This paper describes the design and evaluation of a prototypical information appliance for displaying and exchanging paper and electronic notes and pictures at home. This study addresses two research problems: how to provide minimal and relevant functionality to home users and to investigate novel interaction styles that rely on a combination of computer-vision based technology with commercial pen input devices in order to support easy and learnable interaction. Evaluation of the prototype application suggests that participants were quickly able to use the application and that they liked the idea of combining paper and electronic media.

1. Summary MPEG-4 AVC/AAC has been accepted as an audio-visual encoding format by a number of standards bodies and is poised for wide adoption in IP-based video services over consumer broadband connections (IPTV). In spite of the agreement on the encoding ...

The increase in the bandwidth of wireless channels and the computing power of mobile devices increase the interest in video communications over wireless networks. However, the high error rate and the rapidly changing quality of the radio channels can be devastating for the transport of compressed video. In motion compensated coding, errors due to packet losses are propagated from reference frames to dependant frames causing lasting visual effects. In addition, the bounded playout delay for interactive video limits the effectiveness of retransmission-based error control. In this paper, we propose a mechanism that combines retransmission-based error control with path diversity in wireless networks, to provide different levels of protection to packets according to their importance to the reconstructed video quality. We evaluated the effectiveness of the mechanism under different network conditions. Simulation results show that the mechanism is able to maintain the video quality under different loss rates, with less overhead compared to error control techniques that depend on reference frame updates.

In this paper, a new error concealment algorithm is proposed for the H.264 standard. The algorithm consists of two processes. The first process uses a fuzzy logic method to select the size type of lost blocks. The motion vector of a lost block is calculated from the current frame, if the motion vectors of the neighboring blocks surrounding the lost block are discontinuous. Otherwise, the size type of the lost block can be determined from the preceding frame. The second process is an error concealment algorithm via a proposed adapted multiplereference-frames selection for finding the lost motion vector. The adapted multiple-reference-frames selection is based on the motion estimation analysis of H.264 coding so that the number of searched frames can be reduced. Therefore the most accurate mode of the lost block can be determined with much less computation time in the selection of the lost motion vector. Experimental results show that the proposed algorithm achieves from 0.5 to 4.52 dB improvement when compared to the method in VM 9.0.

Receiver adaptation layered multicast video has been pro- posed to address the issue of network bandwidth heterogeneity in multi- party video communication over networks. An important issue in this ap- proach is how to conduct layer adaptation to achieve the "optimal level" of subscription to maximize the perceived video quality at receivers while uti- lize bandwidth efficiently. In the current work, optimal layers are typically determined by congestion signals fed back from networks. This approach implies that instantaneous congestion is a key parameter; however, in ran- dom traffic, such an implication somewhat under-estimates other parame- ters. In order to investigate the performance of "layer adaptation" in random traffic for improving the network-based multi-party video communication system, in this paper we present an analytical studying of the issue of re- ceiver layer adaptation for layered video transmission. In this work, we use the notion of congestio...

The widespread use of multimedia services on the World Wide Web and the advances in end-user portable devices have recently increased the user demands for better quality. Moreover, providing these services seamlessly and ubiquitously on wireless networks and with user mobility poses hard challenges. To meet these challenges and fulfill the end-user requirements, suitable strategies need to be adopted at both application level and network level. At the application level rate and quality have to be adapted to time-varying bandwidth limitations, whereas on the network side a mechanism for efficient use of the network resources has to be implemented, to provide a better end-user Quality of Experience (QoE) through better Quality of Service (QoS). The work in this thesis addresses these issues by first investigating multi-stream rate adaptation techniques for Scalable Video Coding (SVC) applications aimed at a fair provision of QoE to end-users. Rate Distortion (R-D) models Last but not the least I am thankful to University of Ferrara for giving me the opportunity and funding me during these three years of my Ph.D.

Video streaming over the Internet is a challenging task due, in part to the wide range of bandwidth variations caused by network congestion. To deal with this challenge, we propose an optimal error control scheme for scalable video transmission over the Internet. The three major components of error controlerror resilience, forward error correction (FEC), and error concealment-are considered in the proposed framework. Rate-distortion (RD) optimization is carried out to determine the encoding mode for each packet and the ...

In this paper, we propose an optimal unequal error protection scheme for object-based video communications over differentiated services networks. Our goal is to achieve the best video quality (minimum total expected distortion) with constraints on transmission cost and delay. An end-to-end distortion estimation approach for objectbased video is proposed, which can be used for different packetization schemes. The problem is solved using Lagarangian relaxation and dynamic programming. Experimental results indicate that the proposed unequal error protection schemes can significantly outperform equal error protection methods.

Multimedia applications involving the transmission of video over communication networks are rapidly increasing in popularity. Such applications can greatly benefit from adapting video coding parameters to network conditions as well as adapting network parameters to better support the application requirements. These two dimensions can both be viewed as allocating source and network resources to improve video quality. In this paper, we highlight recent advances in optimal resource allocation for real-time video communications over unreliable and resource constrained communication channels. More specifically, we focus on point-to-point coding and delivery schemes in which the sequences are encoded on the fly. We present a high-level framework for resource-distortion optimization. The framework can be used for jointly considering factors across network layers, including source coding, channel resource allocation, and error concealment. For example, resources can take the form of transmission energy in a wireless channel, and transmission cost in a DiffServ-based Internet channel. This framework can be used to optimally trade off resource consumption with end-to-end video quality in packet-based video transmission. After giving an overview of this framework, we review recent work in two areas-energy efficient wireless video transmission and resource allocation for Internet-based applications.

Object-based video coding is a relatively new technique to meet the fast growing demand for interactive multimedia applications. Compared with conventional frame-based video coding, it consists of two types of source data: shape information and texture information. Recently, joint source-channel coding for multimedia communications has gained increased popularity. However, very limited work has been conducted to address the problem of joint source-channel coding for object-based video. In this paper, we propose a cost-distortion optimal unequal error protection (UEP) scheme for object-based video communications. Our goal is to achieve the best video quality (minimum total expected distortion) with constraints on transmission cost and delay in a lossy network environment. The problem is solved using Lagarangian relaxation and dynamic programming. The performance of the proposed scheme is tested using simulations of a narrow-band block-fading wireless channel with additive white Gaussian noise and a simplified differentiated services Internet channel. Experimental results indicate that the proposed UEP scheme can significantly outperform equal error protection methods. Index Terms-Differentiated service (DiffServ) network, joint source-channel coding, lossy network, MPEG-4 standard, object-based video, rate distortion, unequal error protection (UEP), video coding, video communications, wireless channel. I. INTRODUCTION V IDEO communications over unreliable networks has emerged as an active and challenging area of research and development. A major problem in this area is how to efficiently allocate communication resources in order to achieve the best video quality. For example, in wireless video communications, mobile devices normally have a limited battery supply. This limited energy is consumed in the processing, transmission, and displaying of the video sequence. In this paper, we consider how the average transmission power used by a modulation scheme directly affects the channel characteristics and therefore affects the received video quality. As another example, in a differentiated services (DiffServ) Internet, allocating resources discriminately for aggregated traffic flows is utilized to support Manuscript

There are a large number of applications requiring the compression of video at Very Low Bit Rates (VLBR). Such applications include wireless video conferencing, video over the internet, multimedia database retrieval and remote sensing and monitoring. Recently, the MPEG-4 standardization effort has been a motivating factor to find a solution to this challenging problem. The existing approaches to this problem can generally be grouped into block-based, model-based, and object-oriented. Block-based approaches follow the ...

Hybrid networks consisting of satellite links and mobile ad hoc networks present a series of challenges due to different packet-loss patterns, delay, and, usually, scarce available bandwidth. In this scenario, redundant encoding, in the form of Forward Erasure Correction (FZC) codes [1],[2], can provide an effective protection against losses in multicast videoconferencing and video streaming applications. The use of efficient encoding techniques can provide further reduction on bandwidth requirements. A real test-bed based on a remote video streaming server interconnected via a GEO-satellite pipe to a local wireless LAN is presented in [3], by adopting the multicast network protocol. The satellite pipe is based on the commercial Skyplex network [4] that operates in the Ka-band with the Hot Bird 6 transponder. The developed platform, described in [5], is shown in Figure 8.15. The purpose is to provide users with a low-cost, high-availability platform for performing experiments with I...

In this paper, we address the problem of real-time video streaming over a bandwidth and energy constrained wireless sensor network (WSN) from a small number of dispersed video-sensor nodes (VNs) to a sink by combining forward error correction (FEC) coding with a novel multipath rouing scheme called directional geographical routing (DGR). DGR constructs an application-specific number of multiple disjointed paths for a VN to transmit parallel FEC-protected H.26L real-time video streams over a bandwidth-limited, unreliable networking environment. The multiple paths in DGR facilitate load balancing, bandwidth aggregation, and fast packet delivery. Extensive simulation experiments over randomly generated WSNs show that DGR has the following advantages: (i) lower delay, (ii) substantially longer network lifetime, and (iii) a better received video quality. In particular, DGR improves the average video peak signal-tonoise ratio (PSNR) by up to 3dB, compared to a traditional geographic routing scheme.

We present in this paper a reliable optimized transmission of H.264 video streaming over IEEE 802.11e Wireless Mesh Networks (WMNs) based on a Cross-Layer Adaptive Enhanced Distributed Channel Access (CLA-EDCA) MAC architecture. The IEEE 802.11e EDCA offers a prioritized transmission to guarantee the minimum packets delay and drop rate needed for time bound applications, such as VoIP and Video. However, the standard EDCA scheme does not adapt to the network state to support time critical applications. In order to resolve the problems associated with standard EDCA, we highlight a novel adaptive architecture so as to change the Contention Window ( CW ) after each successful or unsuccessful transmission according to the current network conditions. We have integrated this cross-layer adaptive scheme leveraging smart forward error correction (FEC) schemes through channel state estimation (CSE) to achieve the optimal video transmission adapting to the network conditions. We have also expl...

A transport protocol that supports real-time communication of audio/video frames across campus-area packet switched networks is presented. It is a "best effort" protocol that attempts to ameliorate the effect of jitter, load variation, and packet loss, to provide low latency, synchronized audio and video communications. This goal is realized through four transport and display mechanisms, and a real-time implementation of these mechanisms that integrates operating system services (e.g., scheduling and resource allocation, and device management) with network communication services (e.g., transport protocols), and with application code (e.g., display routines). The four mechanisms are: a facility for varying the synchronization between audio and video to achieve continuous audio in the face of jitter, a network congestion monitoring mechanism that is used to control audio/video latency, a queueing mechanism at the sender that is used to maximize frame throughput without unnecessarily increasing latency, and a forward error correction mechanism for transmitting audio frames multiple times to ameliorate the effects of packet loss in the network. The effectiveness of these techniques is demonstrated by measuring the performance of the protocol when transmitting audio and video across congested networks.

We propose a 2D display and camera arrangement for video communication systems that supports both spatial information between distant sites and user mobility. The implementation of this arrangement is called the "surrounding back screen method." The method enables users to freely come from and go into other users' spaces and provides every user with the direct pointing capability, since there is no apparent spatial barrier separating users, unlike the case of conventional video communication systems. In this paper, we introduce two properties ("sharedness" and "exclusiveness") and three parameters (a distance and two angles) to represent the geometrical relationship between two users. These properties and parameters are used to classify the shared spaces created by a video communication system and to investigate the surrounding back screen method. Furthermore, to demonstrate and explore our surrounding back screen method, we have developed a prototype system, called t-Room. Taking into account practical situations, we studied a practical case where two t-Rooms with different layouts are connected.

We introduce People in Books with FlashCam technology, a system that supports children and long-distance family members to act as characters in children's storybooks while they read stories together over a distance. By segmenting the video chat streams of the child and remote family member from their background surroundings, we create the illusion that the child and adult reader are immersed among the storybook illustrations. The illusion of inhabiting a shared story environment helps remote family members feel a sense of togetherness and encourages active reading behaviors for children ages three to five. People In Books is designed to fit into families' traditional reading practices, such as reading ebooks on couches or in bed via netbook or tablet computers. To accommodate this goal we implemented FlashCam, a computationally cost effective and physically small background subtraction system for mobile devices that allows users to move locations and change lighting conditions while they engage in background-subtracted video communications. A lab evaluation compared People in Books with a conventional remote reading application. Results show that People in Books motivates parents and children to be more performative readers and encourages open-ended play beyond the story, while creating a strong sense of togetherness.

Advancements in educational and communicative technology have provided a new medium with unique properties that bring about new forms of transcultural learning and instruction. Rather than simulating traditional face-to-face classroom methods in online learning environments, the chapter examines different approaches for instructors in creating new ways to link technology to curriculum. This chapter begins by introducing the readers to many of the challenges, concepts, and practices associated with synchronous online learning environments. Through multiple case studies, the chapter will explore the importance of (a) content, pedagogy, and culture, (b) the role of the instructor and student in creating successful online learning environments, and (c) strengths and weaknesses of communication tools and educational technology in creating a blended online learning environment. The chapter helps instructors and students identify the benefits and shortcomings of technology, explore how to enhance dynamic interactions between individuals and groups, and examine how to capitalize on the strengths of audio and visual images to accompany other formats of instructional materials.

In distributed systems, quorum-based algorithms are used to access shared resources in a mutually exclusive way because these algorithms are more resilient to process and communication failures. Based on resource allocation, the quorum system for the mutual exclusion problem is a coterie and the quorum system for the k-mutual exclusion problem is a kcoterie. There are two types of coteries and k-coteries which are dominated and nondominated. The availability of a coterie and a k-coterie need to be considered in an error-prone system. Therefore, k-coterie and coterie join operation is presented to build a new larger k-coterie so that the availability of a k-coterie increases. Through an analysis of availability is obtained that a kcoterie C is nondominated if and only if for any probability p, the sum of the (k,1)-availability and the (k, k)dual availability of a coterie C equals one.

Recent development of depth sensors has facilitated the progress of 2D-plus-depth methods for 3D video representation, for which frame-rate up-conversion (FRUC) of depth video is a critical step. However, due to the computational cost of state-of-the-art FRUC methods, real time applications of 2D-plus-depth is still limited. In this paper, we present a method of speeding up the FRUC of the depth video by treating it as part of a video coding process, combined with a novel color-mapping algorithm is adopted to improve the quality of temporal upsampling. Experiments show that the proposed systems saves up to 99.5% of the frame interpolation time, while achieving virtually identical reconstructed depth video as state-of-the-art methods.

Ubiquitous use of real-time video communication on the Internet requires adaptive applications that can provide different levels of quality depending on the amount of resources available. For video coding this means that the algorithms must be designed to be scalable in terms of bandwidth, processing requirements and quality of the reconstructed signal. This paper presents a novel video compression and coding algorithm targeted at delay-sensitive applications in heterogeneous network and computing environments. The algorithm, based on the embedded zerotree wavelet algorithm for still image compression, generates a highly scalable layered bitstream that can be decoded at different qualities in terms of spatial resolution, frame rate and compression distortion. Furthermore, the algorithm is designed to require only a minimal coding delay, making it suitable for highly interactive communication applications like videoconferencing. The performance of the proposed algorithm is evaluated by comparison with a non- scalable codec and the penalty in compression efficiency that the scalability requirement imposes is analyzed. The codec is shown to produce a scalable bitstream ranging from about 10 kbps to 10 Mbps, while the computational complexity is kept at a level that makes software implementation on CPU-constrained equipment feasible.

In this paper we propose a special bit allocation method which can be used in most rate control algorithms in variable rate video applications. In real-time video communication applications, we need a constant short-term average bitrate, while in variable bitrate applications such as streaming and local recording applications, a constant long-term average bitrate is sufficient and more short-term variation in bitrate is acceptable. In comparison with constant bitrate video, a variable bitrate video can provide better visual quality and coding efficiency for compressed video sequences. Furthermore, while more variation in bitrate is possible we have additional degrees of freedom to control the encoding parameters. We propose a special bit allocation algorithm to take advantage of this freedom in variable bitrate video. We introduce a new type of frame namely SPP frame (SPecial P frame) that can be used in combination with I, P, B and other types of frames in different encoders including H.263, MPEG-4 and H.264/AVC encoders. We propose a simple method to implement the SPP frames independently of the rate control algorithm. The experimental results show that the SPP frames can considerably increase the total average quality of variable rate encoded video.