Adaptive buffer power save mechanism for mobile multimedia streaming

With the proliferation of wireless networks, the use of mobile devices to stream multimedia is growing in popularity. Although the devices are improving in that they are becoming smaller, more complex and capable of running more applications than ever before, there is one aspect of them that is lagging behind. Batteries have seen little development, even though they are one of the most important parts of the devices. Multimedia streaming puts extra pressure on batteries, causing them to discharge faster. This often means that streaming tasks can not be completed, resulting in significant user dissatisfaction. Consequently, effort is required to devise mechanisms to enable and increase in battery life while streaming multimedia. In this context, this thesis presents a novel algorithm to save power in mobile devices during the streaming of multimedia content. The proposed Adaptive-Buffer Power Save Mechanism (AB-PSM) controls how the data is sent over wireless networks, achieving significant power savings. There is little or no effect on the user and the algorithm is very simple to implement. The thesis describes tests which show the effectiveness of AB-PSM in comparison with the legacy power save mechanism present in IEEE 802.11. The thesis also presents a detailed overview of the IEEE 802.11 protocols and an in-depth literature review in the area of power saving during multimedia streaming. A novel analysis of how the battery of a mobile device is affected by multimedia streaming in its different stages is given. A total-power-save algorithm is then described as a possible extension to the Adaptive-Buffer Power Save Mechanism

Quality-Oriented Mobility Management for Multimedia Content Delivery to Mobile Users

The heterogeneous wireless networking environment determined by the latest developments in wireless access technologies promises a high level of communication resources for mobile computational devices. Although the communication resources provided, especially referring to bandwidth, enable multimedia streaming to mobile users, maintaining a high user perceived quality is still a challenging task. The main factors which affect quality in multimedia streaming over wireless networks are mainly the error-prone nature of the wireless channels and the user mobility. These factors determine a high level of dynamics of wireless communication resources, namely variations in throughput and packet loss as well as network availability and delays in delivering the data packets. Under these conditions maintaining a high level of quality, as perceived by the user, requires a quality oriented mobility management scheme. Consequently we propose the Smooth Adaptive Soft-Handover Algorithm, a novel quality oriented handover management scheme which unlike other similar solutions, smoothly transfer the data traffic from one network to another using multiple simultaneous connections. To estimate the capacity of each connection the novel Quality of Multimedia Streaming (QMS) metric is proposed. The QMS metric aims at offering maximum flexibility and efficiency allowing the applications to fine tune the behavior of the handover algorithm. The current simulation-based performance evaluation clearly shows the better performance of the proposed Smooth Adaptive Soft-Handover Algorithm as compared with other handover solutions. The evaluation was performed in various scenarios including multiple mobile hosts performing handover simultaneously, wireless networks with variable overlapping areas, and various network congestion levels

Development of advanced multimedia services in P2P architectures

La transmissió de fluxos multimèdia en temps real (streaming) s’ha convertit en un tema punter i de gran interès al món de la recerca d’Internet, especialment quan ens referim a aplicacions de transmissió d’àudio i vídeo en directe a través de xarxes peer-to-peer (P2P). Generalment, aquestes aplicacions han de fer front a molts problemes en el seu disseny i implementació deguts a la dinamicitat i heterogeneïtat que per natura caracteritzen les xarxes P2P. En aquest projecte, s’introdueixen noves característiques que les aplicacions de transmissió multimèdia P2P actuals no contemplen. Els requisits de connexió i maquinari són diferents per a la transmissió de fluxos de baixa i alta capacitat, no obstant, tots els nodes s’acostumen a considerar idèntics, cosa que no representa una aproximació gaire encertada tenint en compte un medi tan heterogeni. A més a més, amb la finalitat d’aconseguir distinció entre nodes, es fa necessari la introducció d’un mecanisme que permeti l’intercanvi de les capacitats específiques de cada node, incloent-hi les de transcodificació de fluxos. Un altre aspecte a destacar és el fet que aquestes aplicacions són difícils d’ampliar, incorporar nous serveis o modificar les dades que porten precarregades com ara la llista de canals de televisió disponibles, cosa que impossibilita garantir la disponibilitat de la font tot el temps. Per altra banda, els serveis interactius tampoc s’han desenvolupat gaire. Aquest projecte proposa el disseny i implementació d’una plataforma de difusió multimèdia P2P cooperativa i interactiva que permet superar els problemes esmentats. La plataforma integra diferents mecanismes que permeten la distribució en temps real de continguts multimèdia en diferents qualitats incloent fluxos d’alta capacitat (com per exemple HD). Aquesta plataforma és una solució novedosa basada en JXTA, DONET i ALM (Arbres Multicast a nivell d’Aplicació) que proporciona un sistema ampliable segons noves necessitats i facilita la inserció de nous serveis de valor afegit. La plataforma proposada es fonamenta en la creació d’una arquitectura de 2 capes lògiques superposades: una capa lògica JXTA, encarregada bàsicament de la senyalització i intercanvi de metadades, i una capa de transmissió basada en sockets UDP unicast. D’aquesta manera, la diferència entre la capa de transmissió i la capa física es pot veure reduïda a partir de la informació obtinguda de la capa JXTA, la qual es va actualitzant al llarg del temps

Computational inference and control of quality in multimedia services

Quality is the degree of excellence we expect of a service or a product. It is also one of the key factors that determine its value. For multimedia services, understanding the experienced quality means understanding how the delivered delity, precision and reliability correspond to the users' expectations. Yet the quality of multimedia services is inextricably linked to the underlying technology. It is developments in video recording, compression and transport as well as display technologies that enables high quality multimedia services to become ubiquitous. The constant evolution of these technologies delivers a steady increase in performance, but also a growing level of complexity. As new technologies stack on top of each other the interactions between them and their components become more intricate and obscure. In this environment optimizing the delivered quality of multimedia services becomes increasingly challenging. The factors that aect the experienced quality, or Quality of Experience (QoE), tend to have complex non-linear relationships. The subjectively perceived QoE is hard to measure directly and continuously evolves with the user's expectations. Faced with the diculty of designing an expert system for QoE management that relies on painstaking measurements and intricate heuristics, we turn to an approach based on learning or inference. The set of solutions presented in this work rely on computational intelligence techniques that do inference over the large set of signals coming from the system to deliver QoE models based on user feedback. We furthermore present solutions for inference of optimized control in systems with no guarantees for resource availability. This approach oers the opportunity to be more accurate in assessing the perceived quality, to incorporate more factors and to adapt as technology and user expectations evolve. In a similar fashion, the inferred control strategies can uncover more intricate patterns coming from the sensors and therefore implement farther-reaching decisions. Similarly to natural systems, this continuous adaptation and learning makes these systems more robust to perturbations in the environment, longer lasting accuracy and higher eciency in dealing with increased complexity. Overcoming this increasing complexity and diversity is crucial for addressing the challenges of future multimedia system. Through experiments and simulations this work demonstrates that adopting an approach of learning can improve the sub jective and objective QoE estimation, enable the implementation of ecient and scalable QoE management as well as ecient control mechanisms

A novel multimedia adaptation architecture and congestion control mechanism designed for real-time interactive applications

PhDThe increasing use of interactive multimedia applications over the Internet has created a problem of congestion. This is because a majority of these applications do not respond to congestion indicators. This leads to resource starvation for responsive flows, and ultimately excessive delay and losses for all flows therefore loss of quality. This results in unfair sharing of network resources and increasing the risk of network ‘congestion collapse’. Current Congestion Control Mechanisms such as ‘TCP-Friendly Rate Control’ (TFRC) have been able to achieve ‘fair-share’ of network resource when competing with responsive flows such as TCP, but TFRC’s method of congestion response (i.e. to reduce Packet Rate) is not ideally matched for interactive multimedia applications which maintain a fixed Frame Rate. This mismatch of the two rates (Packet Rate and Frame Rate) leads to buffering of frames at the Sender Buffer resulting in delay and loss, and an unacceptable reduction of quality or complete loss of service for the end-user. To address this issue, this thesis proposes a novel Congestion Control Mechanism which is referred to as ‘TCP-friendly rate control – Fine Grain Scalable’ (TFGS) for interactive multimedia applications. This new approach allows multimedia frames (data) to be sent as soon as they are generated, so that the multimedia frames can reach the destination as quickly as possible, in order to provide an isochronous interactive service. This is done by maintaining the Packet Rate of the Congestion Control Mechanism (CCM) at a level equivalent to the Frame Rate of the Multimedia Encoder.The response to congestion is to truncate the Packet Size, hence reducing the overall bitrate of the multimedia stream. This functionality of the Congestion Control Mechanism is referred to as Packet Size Truncation (PST), and takes advantage of adaptive multimedia encoding, such as Fine Grain Scalable (FGS), where the multimedia frame is encoded in order of significance, Most to Least Significant Bits. The Multimedia Adaptation Manager (MAM) truncates the multimedia frame to the size indicated by the Packet Size Truncation function of the CCM, accurately mapping user demand to available network resource. Additionally Fine Grain Scalable encoding can offer scalability at byte level granularity, providing a true match to available network resources. This approach has the benefits of achieving a ‘fair-share’ of network resource when competing with responsive flows (as similar to TFRC CCM), but it also provides an isochronous service which is of crucial benefit to real-time interactive services. Furthermore, results illustrate that an increased number of interactive multimedia flows (such as voice) can be carried over congested networks whilst maintaining a quality level equivalent to that of a standard landline telephone. This is because the loss and delay arising from the buffering of frames at the Sender Buffer is completely removed. Packets sent maintain a fixed inter-packet-gap-spacing (IPGS). This results in a majority of packets arriving at the receiving end at tight time intervals. Hence, this avoids the need of using large Playout (de-jitter) Buffer sizes and adaptive Playout Buffer configurations. As a result this reduces delay, improves interactivity and Quality of Experience (QoE) of the multimedia application

Video Quality Prediction for Video over Wireless Access Networks (UMTS and WLAN)

Transmission of video content over wireless access networks (in particular, Wireless Local Area Networks (WLAN) and Third Generation Universal Mobile Telecommunication System (3G UMTS)) is growing exponentially and gaining popularity, and is predicted to expose new revenue streams for mobile network operators. However, the success of these video applications over wireless access networks very much depend on meeting the user’s Quality of Service (QoS) requirements. Thus, it is highly desirable to be able to predict and, if appropriate, to control video quality to meet user’s QoS requirements. Video quality is affected by distortions caused by the encoder and the wireless access network. The impact of these distortions is content dependent, but this feature has not been widely used in existing video quality prediction models. The main aim of the project is the development of novel and efficient models for video quality prediction in a non-intrusive way for low bitrate and resolution videos and to demonstrate their application in QoS-driven adaptation schemes for mobile video streaming applications. This led to five main contributions of the thesis as follows:(1) A thorough understanding of the relationships between video quality, wireless access network (UMTS and WLAN) parameters (e.g. packet/block loss, mean burst length and link bandwidth), encoder parameters (e.g. sender bitrate, frame rate) and content type is provided. An understanding of the relationships and interactions between them and their impact on video quality is important as it provides a basis for the development of non-intrusive video quality prediction models.(2) A new content classification method was proposed based on statistical tools as content type was found to be the most important parameter. (3) Efficient regression-based and artificial neural network-based learning models were developed for video quality prediction over WLAN and UMTS access networks. The models are light weight (can be implemented in real time monitoring), provide a measure for user perceived quality, without time consuming subjective tests. The models have potential applications in several other areas, including QoS control and optimization in network planning and content provisioning for network/service providers.(4) The applications of the proposed regression-based models were investigated in (i) optimization of content provisioning and network resource utilization and (ii) A new fuzzy sender bitrate adaptation scheme was presented at the sender side over WLAN and UMTS access networks. (5) Finally, Internet-based subjective tests that captured distortions caused by the encoder and the wireless access network for different types of contents were designed. The database of subjective results has been made available to research community as there is a lack of subjective video quality assessment databases.Partially sponsored by EU FP7 ADAMANTIUM Project (EU Contract 214751

Quality of experience in digital mobile multimedia services

People like to consume multimedia content on mobile devices. Mobile networks can deliver mobile TV services but they require large infrastructural investments and their operators need to make trade-offs to design worthwhile experiences. The approximation of how users experience networked services has shifted from the inadequate packet level Quality of Service (QoS) to the user perceived Quality of Experience (QoE) that includes content, user context and their expectations. However, QoE is lacking concrete operationalizations for the visual experience of content on small, sub-TV resolution screens displaying transcoded TV content at low bitrates. The contribution of my thesis includes both substantive and methodological results on which factors contribute to the QoE in mobile multimedia services and how. I utilised a mix of methods in both lab and field settings to assess the visual experience of multimedia content on mobile devices. This included qualitative elicitation techniques such as 14 focus groups and 75 hours of debrief interviews in six experimental studies. 343 participants watched 140 hours of realistic TV content and provided feedback through quantitative measures such as acceptability, preferences and eye-tracking. My substantive findings on the effects of size, resolution, text quality and shot types can improve multimedia models. My substantive findings show that people want to watch mobile TV at a relative size (at least 4cm of screen height) similar to living room TV setups. In order to achieve these sizes at 35cm viewing distance users require at least QCIF resolution and are willing to scale it to a much lower angular resolution (12ppd) then what video quality research has found to be the best visual quality (35ppd). My methodological findings suggest that future multimedia QoE research should use a mixed methods approach including qualitative feedback and viewing ratios akin to living room setups to meet QoE’s ambitious scope

Robust P2P Live Streaming

Projecte fet en col.laboració amb la Fundació i2CATThe provisioning of robust real-time communication services (voice, video, etc.) or media contents through the Internet in a distributed manner is an important challenge, which will strongly influence in current and future Internet evolution. Aware of this, we are developing a project named Trilogy leaded by the i2CAT Foundation, which has as main pillar the study, development and evaluation of Peer-to-Peer (P2P) Live streaming architectures for the distribution of high-quality media contents. In this context, this work concretely covers media coding aspects and proposes the use of Multiple Description Coding (MDC) as a flexible solution for providing robust and scalable live streaming over P2P networks. This work describes current state of the art in media coding techniques and P2P streaming architectures, presents the implemented prototype as well as its simulation and validation results