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.

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.

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.

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.

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.

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.

Protection of images that are encoded using subband coding from channel error is addressed. In this scheme the low-pass subband is encoded using DPCM (differential pulse-code modulation), and the other subbands are encoded using a scalar quantizer. The quantizers are all Lloyd-Max quantizers, from which the representation levels have fixed length codewords. First, considering only single errors in each codeword, a channel error distortion measure is derived for each quantizer, that is, for each subband. Codewords are assigned to the quantizer representation levels, yielding a low value of the distortion measure. Next, sets S/sub ij/ consisting of the jth bit from subband i are formed. Each set S/sub ij/ is assigned a particular BCH code C/sub ij/. An algorithm that optimally assigns BCH codes C/sub ij/ to each set S/sub ij/, based on a channel error distortion measure for the entire image, is derived. The protection scheme is adaptive, because each set of bits within each subband can be assigned a different error protection code. Examples show that this approach is preferable to assigning equal error protection codes to each set of bits. It is shown that in the case of a channel error probability of 10/sup -3/, only 5% to 10% extra bits are needed for adequate channel error protection.>

The authors investigate the application of subband coding (SBC) to the progressive transmission of images. The main objective of progressive transmission is to allow the receiver to recognize a picture as quickly as possible at minimum cost. After this it can be decided to receive further picture details or to abort the transmission. SBC turns out to be very suitable for such a system, because the low-pass version of the image is easily recognized for any image type. Further details of the picture are obtained by sequentially receiving the (encoded) higher bandpass versions of the image. Because SBC has good data compression properties, not many bits are needed to transmit the low-pass image and the other subbands. Comparisons to other progressive transmission schemes, such as 8*8 block DCT (discrete cosine transform) encoding, show that SBC is favorable, both in mean-squared error and in a subjective sense.>

An optimal bit allocation algorithm is presented that is suitable for all practical situations. Each source to be coded is assumed to have its own set of admissible quantizers (which can be either scalar or vector quantizers) which do not need to have integer bit rates. The distortion versus rate characteristic of each quantizer set may have an arbitrary shape. The algorithm is very simple in structure and can be applied to any practical coding scheme (such as a subband coder) that needs dynamic bit allocation.>

A novel two-dimensional subband coding technique is presented that can be applied to images as well as speech. A frequency-band decomposition of the image is carried out by means of 2D separable quadrature mirror filters, which split the image spectrum into 16 equal-rate subbands. These 16 parallel subband signals are regarded as a 16-dimensional vector source and coded as such using vector quantization. In the asymptotic case of high bit rates, a theoretical analysis yields that a lower bound to the gain is attainable by choosing this approach over scalar quantization of each subband with an optimal bit allocation. It is shown that vector quantization in this scheme has several advantages over coding the subbands separately. Experimental results are given, and it is shown the scheme has a performance that is comparable to that of more complex coding techniques.>

This paper presents a form of sub-band coding of images in which the advantages of both DPCM and vector quantization are combined in a single subband coder. A 16 band decomposition of the image frequency band is employed using a tree structure of two-dimensional separable quadrature mirror filters and decimators. The 16 parallel sub-bands created in this way are regarded as a 16-dimensional vector process and coded as such using a vector equivalent of DPCM. Here each vector is predicted as a single quantity and its residual is coded using vector quantization, this vector quantizer being nested within the vector-DPCM loop. As will be shown this technique yields a good SNR performance in the bit rate region between 0.4 and 0.7 bits per pixel, both for images inside and outside a large training set.