Peter Westerink

Two-pass MPEG-2 variable-bit-rate encoding

Many MPEG-2 encoding applications are real-time; this implies that the video signal must be encoded with no significant lookahead. However, there exist non-real-time applications that do enable us to first analyze a video sequence entirely, and, using the analysis results, to optimize a second encoding pass of the same data. One example of such an application is the digital video disk (DVD), which is designed to facilitate a variable-bit-rate (VBR) output stream. In that case, it is possible to let the MPEG-2 encoder produce a video sequence with a constant visual quality over time. This is in contrast to constant-bit-rate (CBR) systems, where the rate is constant but the visual quality varies with the coding difficulty. This paper describes a two-pass encoding system that has as its objective to produce an optimized VBR data stream in a second pass. In a first pass, the video sequence is encoded with CBR, while statistics concerning coding complexity are gathered. Next, the first-pass data is processed to prepare the control parameters for the second pass, which performs the actual VBR compression. In this off-line processing stage, we determine the target number of bits for each picture in the sequence, such that we realize the VBR objective. This means that the available bits are appropriately distributed over the different video segments such that constant visual quality is obtained. To be able to quantify the constant visual quality, perceptual experiments are described and a practical model is fitted to them. Exceptional cases such as scene changes and fades are detected and dealt with appropriately. We also ensure that the second-pass compression process does not violate the decoder buffer boundaries. Finally, the encoding is performed again, but now under control of the processed first-pass data. During the running of this second pass, a run-time bit-production control mechanism monitors the accuracy and validity of the first-pass data, correcting errors in prediction and observing the buffer boundaries. Results are compared to CBR operation.

Securing media for adaptive streaming

This paper describes the ARMS system which enables secure and adaptive rich media streaming to a large-scale, heterogeneous client population. The secure streaming algorithms ensure end-to-end security while the content is adapted and streamed via intermediate, potentially untrusted servers. ARMS streaming is completely standards compliant and to our knowledge is the first such end-to-end MPEG-4-based system.

Low-delay MPEG-2 video coding

High-quality and low-delay MPEG-2 video coding can be achieved by avoiding the use of intra (I) and bidirectional prediction (B) pictures. Such coding requires intra macroblocks refreshing techniques for channel error propagation resilience and for compliance with the accuracy requirements of the MPEG-2 standard. This paper describes some of these techniques and presents software simulation results of their performance in terms of image quality and their robustness of transmission channel errors.

ARMS: adaptive rich media secure streaming

In this demonstration we present the ARMS system which enables secure and adaptive rich media streaming to a large-scale, heterogeneous client population. The ARMS system dynamically adapts streams to available bandwidth, client capabilities, packet loss, and administratively imposed policies - all while maintaining full content security. The ARMS system is completely standards compliant and to our knowledge is the first such end-to-end MPEG-4-based system.

Orthogonal layered multicast: improving the multicast transmission of multimedia streams at multiple data rates

This paper discusses orthogonality in layered multicast schemes, and presents a general framework for the implementation of layered multicast. This framework can be applied to standard IP multicast or to any other multicast technology capable of transmitting a data stream to a group of subscriber nodes. The multirate transmission is accomplished by a multicast group management facility which maps the frames of the multimedia stream into a number of multicast groups according to different data resolutions. The subscriber nodes can then directly control the data rate of the stream that they receive by joining different multicast groups, thus eliminating the need for constant data rate negotiation with the sender node.

Polyglot Application Auto Scaling Service for Platform as a Service Cloud

Platform as a service (PaaS) is a cloud delivery model that provides software services and solution stacks to enable rapid development, deployment, and operations in many languages and run-times (polyglot). These applications require capabilities to rapidly grow and shrink the underlying resources to satisfy their workload needs. Auto scaling is a service that enables dynamic resource allocation and deal location to match application performance needs and service level agreements. In this paper we present the architecture and implementation of a polyglot auto scaling solution for IBM Blue mix PaaS. Our auto scaling service enables users to describe policies and set thresholds for scaling the applications based on CPU, memory and heap usage for applications developed in different languages (Java, Java Script, Ruby, etc). The auto scaling service consists of a set of monitoring agents, monitoring service, scaling service, and a persistence service. The service is developed with sharedmulti-tenancy model and offered as a managed cloud service. An application attached to the auto scaling service is monitored and its resources will be adjusted based on the auto scaling policies of the user and on the system conditions.

Non-intrusive Adaptive Multi-media Routing in Peer-to-Peer Multi-party Video Conferencing

Motivated by the problem of limited bandwidth in peer-to-peer (P2P) multi-party video conferencing systems, in this paper we propose a non-intrusive adaptive multi-media routing algorithm that effectively calculates stream routing to achieve a maximum number of receiving streams. The technique is non-intrusive in that it makes use of current streaming status to infer link bottlenecks rather than sending active probing packets, which would seriously interfere with the latency-sensitive video conferencing application and waste bandwidth. When link bottlenecks are detected, the method will adaptively calculate streaming routes, allowing bandwidth abundant peers to act as relays. To test the performance, we use real data from a world wide bandwidth distribution archive and investigate the algorithm convergence rate and distribution fairness through simulation. Results show that the technique works well to achieve effective multi-media routing for latency-sensitive video conferencing applications.

A live intranet distance learning system using MPEG-4 over RTP/RTSP

This paper describes an application where multiple live audio/visual data streams are combined in MPEG-4 and streamed via RTP/RTSP. The scenario involves the video and audio of a classroom instructor, where the overhead foils the instructor may use are sent as an additional separate data stream. The three streams are encapsulated in MPEG-4 systems, which adds synchronization among the streams and a combined composition (positioning and sizing) into a single multimedia presentation. The MPEG-4 classroom presentation is multicast live using RTP/RTSP, which allows remote students to join the class at any time. Additionally, the live presentation is captured in an MPEG-4 file. This will allow for repeating the same class again over RTP/RTSP, or for later review via HTTP on request. The client is an MPEG-4 player written in pure Java, whereas the server is an extension of an existing IBM video server. The system has been implemented and a demonstration is given.

A high level flexible framework for building multi-platform multi-media streaming applications

In this paper we use multi-media streaming as a practical example of how one could design a system for maximum re-use and present the implications of using this approach. While the idea of re-use is of course widely used, we found that many multi-media applications are dedicated to the specific task it was designed for and only the components of the application seem to get re-used in other applications. A lot of effort is expended over and over again to re-use just those components in building new applications. To address this problem, we have developed a high level software framework for building multi-media applications, which makes it easy to build, maintain, and extend a variety of multi-media and emerging collaboration applications. One reason for the ease-of-use is that the framework is at such a level that the intricacies and details of multi-media streaming and processing need not be explicitly known to the integrator, who does not have to worry about how to connect the components, control the timing and threading, and such. Examples of possible applications that can be built using this new framework include a player, a Video-on-Demand server, a video conferencing client, a multi-media presentation recorder, a trans-coder, a protocol translator, etc. For maximum applicability the software is platform independent, has pluggable components such as encoders and decoders, and most modules adhere to multi-media standards from the IETF and MPEG. We will show the ease of integration and maintenance and give demonstrations of some of the projects and products in our company that use this framework, such as collaboration recording, audio/visual help desk support, one-to-many presentations of audio/video with synchronized slides, peer-to-peer video conferencing and collaboration, and speech-to-text as a remote real-time service.

A Feasibility Study of Collaborative Stream Routing in Peer-to-Peer Multiparty Video Conferencing

Video transmission in multiparty video conferencing is challenging due to the demanding bandwidth usage and stringent latency requirement. In this paper, we systematically analyze the problem of collaborative stream routing using one-hop forwarding assistance in a bandwidth constraint environment. We model the problem as a multi-source degree-constrained multicast tree construction problem, and investigate heuristic algorithms to construct bandwidth-feasible shared multicast trees. The contribution of this work is primarily two-fold: (1) we study the solution space of finding a feasible bandwidth configuration for stream routing in a peer-to-peer (P2P) setting, and propose two heuristic algorithms that can quickly produce a bandwidth-feasible solution, making them suitable for large-scale conference sessions, (2) we conduct an empirical study using a realistic dataset and show the effectiveness of our heuristic algorithms. Various QoS metrics are taken into account to evaluate the performance of our algorithms. Finally, we discuss open issues for further exploration. The feasibility study presented in this paper will shed light on the design and implementation of practical P2P multiparty video conferencing applications.