Digital television applications

Studying development of interactive services for digital television is a leading edge area of work as there is minimal research or precedent to guide their design. Published research is limited and therefore this thesis aims at establishing a set of computing methods using Java and XML technology for future set-top box interactive services. The main issues include middleware architecture, a Java user interface for digital television, content representation and return channel communications. The middleware architecture used was made up of an Application Manager, Application Programming Interface (API), a Java Virtual Machine, etc., which were arranged in a layered model to ensure the interoperability. The application manager was designed to control the lifecycle of Xlets; manage set-top box resources and remote control keys and to adapt the graphical device environment. The architecture of both application manager and Xlet forms the basic framework for running multiple interactive services simultaneously in future set-top box designs. User interface development is more complex for this type of platform (when compared to that for a desktop computer) as many constraints are set on the look and feel (e.g., TV-like and limited buttons). Various aspects of Java user interfaces were studied and my research in this area focused on creating a remote control event model and lightweight drawing components using the Java Abstract Window Toolkit (AWT) and Java Media Framework (JMF) together with Extensible Markup Language (XML). Applications were designed aimed at studying the data structure and efficiency of the XML language to define interactive content. Content parsing was designed as a lightweight software module based around two parsers (i.e., SAX parsing and DOM parsing). The still content (i.e., text, images, and graphics) and dynamic content (i.e., hyperlinked text, animations, and forms) can then be modeled and processed efficiently. This thesis also studies interactivity methods using Java APIs via a return channel. Various communication models are also discussed that meet the interactivity requirements for different interactive services. They include URL, Socket, Datagram, and SOAP models which applications can choose to use in order to establish a connection with the service or broadcaster in order to transfer data. This thesis is presented in two parts: The first section gives a general summary of the research and acts as a complement to the second section, which contains a series of related publications.reviewe

Distributed multimedia systems

A distributed multimedia system (DMS) is an integrated communication, computing, and information system that enables the processing, management, delivery, and presentation of synchronized multimedia information with quality-of-service guarantees. Multimedia information may include discrete media data, such as text, data, and images, and continuous media data, such as video and audio. Such a system enhances human communications by exploiting both visual and aural senses and provides the ultimate flexibility in work and entertainment, allowing one to collaborate with remote participants, view movies on demand, access on-line digital libraries from the desktop, and so forth. In this paper, we present a technical survey of a DMS. We give an overview of distributed multimedia systems, examine the fundamental concept of digital media, identify the applications, and survey the important enabling technologies.published_or_final_versio

Dynamic adaptation of streamed real-time E-learning videos over the internet

Even though the e-learning is becoming increasingly popular in the academic environment, the quality of synchronous e-learning video is still substandard and significant work needs to be done to improve it. The improvements have to be brought about taking into considerations both: the network requirements and the psycho- physical aspects of the human visual system. One of the problems of the synchronous e-learning video is that the head-and-shoulder video of the instructor is mostly transmitted. This video presentation can be made more interesting by transmitting shots from different angles and zooms. Unfortunately, the transmission of such multi-shot videos will increase packet delay, jitter and other artifacts caused by frequent changes of the scenes. To some extent these problems may be reduced by controlled reduction of the quality of video so as to minimise uncontrolled corruption of the stream. Hence, there is a need for controlled streaming of a multi-shot e-learning video in response to the changing availability of the bandwidth, while utilising the available bandwidth to the maximum. The quality of transmitted video can be improved by removing the redundant background data and utilising the available bandwidth for sending high-resolution foreground information. While a number of schemes exist to identify and remove the background from the foreground, very few studies exist on the identification and separation of the two based on the understanding of the human visual system. Research has been carried out to define foreground and background in the context of e-learning video on the basis of human psychology. The results have been utilised to propose methods for improving the transmission of e-learning videos. In order to transmit the video sequence efficiently this research proposes the use of Feed- Forward Controllers that dynamically characterise the ongoing scene and adjust the streaming of video based on the availability of the bandwidth. In order to satisfy a number of receivers connected by varied bandwidth links in a heterogeneous environment, the use of Multi-Layer Feed-Forward Controller has been researched. This controller dynamically characterises the complexity (number of Macroblocks per frame) of the ongoing video sequence and combines it with the knowledge of availability of the bandwidth to various receivers to divide the video sequence into layers in an optimal way before transmitting it into network. The Single-layer Feed-Forward Controller inputs the complexity (Spatial Information and Temporal Information) of the on-going video sequence along with the availability of bandwidth to a receiver and adjusts the resolution and frame rate of individual scenes to transmit the sequence optimised to give the most acceptable perceptual quality within the bandwidth constraints. The performance of the Feed-Forward Controllers have been evaluated under simulated conditions and have been found to effectively regulate the streaming of real-time e-learning videos in order to provide perceptually improved video quality within the constraints of the available bandwidth

Study and simulation of low rate video coding schemes

The semiannual report is included. Topics covered include communication, information science, data compression, remote sensing, color mapped images, robust coding scheme for packet video, recursively indexed differential pulse code modulation, image compression technique for use on token ring networks, and joint source/channel coder design

SatCom Today in Canada: Significant Research: Broadband Satellite Communications List of CITR related Publications (1998-2003)

Journal Papers Conference Papers Contributions to Standards Canadian Space Agency Recent Publication

Reconfigurable media coding: a new specification model for multimedia coders

Multimedia coding technology, after about 20 years of active research, has delivered a rich variety of different and complex coding algorithms. Selecting an appropriate subset of these algorithms would, in principle, enable a designer to produce the codec supporting any desired functionality as well as any desired trade-off between compression performance and implementation complexity. Currently, interoperability demands that this selection process be hard-wired into the normative descriptions of the codec, or at a lower level, into a predefined number of choices, known as profiles, codified within each standard specification. This paper presents an alternative paradigm for codec deployment that is currently under development by MPEG, known as Reconfigurable Media Coding (RMC). Using the RMC framework, arbitrary combinations of fundamental algorithms may be assembled, without predefined standardization, because everything necessary for specifying the decoding process is delivered alongside the content itself. This side-information consists of a description of the bitstream syntax, as well as a description of the decoder configuration. Decoder configuration information is provided as a description of the interconnections between algorithmic blocks. The approach has been validated by development of an RMC format that matches MPEG-4 Video, and then extending the format by adding new chroma-subsampling patterns

Performance analysis of an ATM network with multimedia traffic: a simulation study

Traffic and congestion control are important in enabling ATM networks to maintain the Quality of Service (QoS) required by end users. A Call Admission Control (CAC) strategy ensures that the network has sufficient resources available at the start of each call, but this does not prevent a traffic source from violating the negotiated contract. A policing strategy (User Parameter Control (UPC)) is also required to enforce the negotiated rates for a particular connection and to protect conforming users from network overload. The aim of this work is to investigate traffic policing and bandwidth management at the User to Network Interface (UNI). A policing function is proposed which is based on the leaky bucket (LB) which offers improved performance for both real time (RT) traffic such as speech and video and non-real time (non-RT) traffic, mainly data by taking into account the QoS requirements. A video cell in violation of the negotiated bit rate causes the remainder of the slice to be discarded. This 'tail clipping' provides protection for the decoder from damaged video slices. Speech cells are coded using a frequency domain coder, which places the most significant bits of a double speech sample into a high priority cell and the least significant bits into a high priority cell. In the case of congestion, the low priority cell can be discarded with little impact on the intelligibility of the received speech. However, data cells require loss-free delivery and are buffered rather than being discarded or tagged for subsequent deletion. This triple strategy is termed the super leaky bucket (SLB). Separate queues for RT and non-RT traffic, are also proposed at the multiplexer, with non pre-emptive priority service for RT traffic if the queue exceeds a predetermined threshold. If the RT queue continues to grow beyond a second threshold, then all low priority cells (mainly speech) are discarded. This scheme protects non-RT traffic from being tagged and subsequently discarded, by queueing the cells and also by throttling back non-RT sources during periods of congestion. It also prevents the RT cells from being delayed excessively in the multiplexer queue. A simulation model has been designed and implemented to test the proposal. Realistic sources have been incorporated into the model to simulate the types of traffic which could be expected on an ATM network. The results show that the S-LB outperforms the standard LB for video cells. The number of cells discarded and the resulting number of damaged video slices are significantly reduced. Dual queues with cyclic service at the multiplexer also reduce the delays experienced by RT cells. The QoS for all categories of traffic is preserved

Semi-synchronous video for deaf telephony with an adapted synchronous codec

Magister Scientiae - MScCommunication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people. Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf people preferred to communicate with both Deaf and hearing peers in South African Sign Language as opposed to text. Synchronous video chat and video relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in developing countries like South Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video codecs to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the expense of some additional latency. Synchronous video codec adaptation consisted of comparing codecs, and choosing one to optimise in order to minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no such standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality metrics with subjective assessment metrics conducted with the local Deaf community.South Afric

Synthesizing hardware from dataflow programs: An MPEG-4 simple profile decoder case study

International audienceThe MPEG Reconfigurable Video Coding working group is developing a new library-based process for building the reference codecs of future MPEG standards, which is based on dataflow and uses an actor language called CAL. The paper presents a code generator producing RTL targeting FPGAs for CAL, outlines its structure, and demonstrates its performance on an MPEG-4 Simple Profile decoder. The resulting implementation is smaller and faster than a comparable RTL reference design, and the second half of the paper discusses some of the reasons for this counter-intuitive result