With the latest developments in video coding technology and fast deployment of end-user broadband internet connections, real-time media applications become increasingly interesting for both private users and businesses. However, the internet remains a best-effort service network unable to guarantee the stringent requirements of the media application, in terms of high, constant bandwidth, low packet loss rate and transmission delay. Therefore, efficient adaptation mechanisms must be derived in order to bridge the application requirements with the transport medium characteristics. Lately, different network architectures, e.g., peer-to-peer networks, content distribution networks, parallel wireless services, emerge as potential solutions for reducing the cost of communication or infrastructure, and possibly improve the application performance. In this thesis, we start from the path diversity characteristic of these architectures, in order to build a new framework, specific for media streaming in multipath networks. Within this framework we ...

With the joint development of networking and digital coding technologies multimedia and more particularly video services are clearly becoming one of the major consumers of the new information networks. The rapid growth of the Internet and computer industry however results in a very heterogeneous infrastructure commonly overloaded. Video service providers have nevertheless to oer to their clients the best possible quality according to their respective capabilities and communication channel status. The Quality of Service is not only inuenced by the compression artifacts, but also by unavoidable packet losses. Hence, the packet video stream has clearly to fulll possibly contradictory requirements, that are coding eciency and robustness to data loss. The rst contribution of this thesis is the complete modeling of the video Quality of Service (QoS) in standard and more particularly MPEG-2 applications. The performance of Forward Error Control (FEC) ...

This thesis presents a comprehensive study of the evaluation of the Quality of Experience (QoE) perceived by the users of 3D video systems, analyzing the impact of effects introduced by all the elements of the 3D video processing chain. Therefore, various subjective assessment tests are presented, particularly designed to evaluate the systems under consideration, and taking into account all the perceptual factors related to the 3D visual experience, such as depth perception and visual discomfort. In particular, a subjective test is presented, based on evaluating typical degradations that may appear during the content creation, for instance due to incorrect camera calibration or video processing algorithms (e.g., 2D to 3D conversion). Moreover, the process of generation of a high-quality dataset of 3D stereoscopic videos is described, which is freely available for the research community, and has been already widely used in ...

Provisioning of digital video services is a difficult task as it is hard to estimate optimal settings of video parameters, given transmission constraints, while maximizing the overall end-user quality. With Internet streaming services becoming part of our everyday life, end-to-end optimization of such systems is important. On one hand, huge effort is given into subjective or objective evaluation of the end-user perception. High quality audiovisual perception with respect to the minimized costs of the provided service is one of the main interests for the network providers. On the other hand, subjective evaluations to determine best video and audio configurations are often evaluated in controlled test laboratory environments, which have little to do with the real environments in which consumers enjoy such content. Unfortunately, no serious attempts have been made to take into account interactions between quality of the content and ...

In this thesis, we propose Multiple Objective Optimization (MOO) frameworks for efficient video streaming. Firstly, we introduce pre-roll delay-distortion optimization (DDO) for uninterrupted content-adaptive video streaming over low capacity, constant bitrate (CBR) channels using MOO. Content analysis is used to divide the input video into shots with assigned relevance levels. The video is adaptively encoded and streamed aiming minimum pre-roll delay and distortion with the optimal spatial and temporal resolutions and quantization parameters for each shot. With buffer and distortion constraints, the bitrate of unimportant shots is reduced to achieve an acceptable quality in important shots. Secondly, we introduce a cross-layer optimized video rate adaptation and scheduling scheme to achieve maximum "application layer" Quality-of-Service (QoS), maximum video throughput (video seconds per transmission slot), and QoS fairness for wireless video streaming. Using the MOO framework, these objectives are jointly optimized such ...

This Ph.D. thesis focuses on developing a system for high-quality speech synthesis and voice conversion. Vocoder-based speech analysis, manipulation, and synthesis plays a crucial role in various kinds of statistical parametric speech research. Although there are vocoding methods which yield close to natural synthesized speech, they are typically computationally expensive, and are thus not suitable for real-time implementation, especially in embedded environments. Therefore, there is a need for simple and computationally feasible digital signal processing algorithms for generating high-quality and natural-sounding synthesized speech. In this dissertation, I propose a solution to extract optimal acoustic features and a new waveform generator to achieve higher sound quality and conversion accuracy by applying advances in deep learning. The approach remains computationally efficient. This challenge resulted in five thesis groups, which are briefly summarized below. I introduce firstly a new method to shape the ...

This thesis aims at contributing to overcome two of the main challenges for the deployment of highly dense wireless M2M networks in data collection scenarios for the Internet of Things: the management of massive numbers of end-devices that attempt to get access to the wireless channel; and the need to extend the network lifetime to reduce maintenance costs. In order to solve these challenges, two complementary strategies are considered. Firstly, the thesis focuses on the design, analysis and performance evaluation of random and hybrid access protocols that can handle abrupt transitions in the traffic load and minimize the energy consumption devoted to communications. And secondly, the use of energy harvesting (EH) is considered in order to provide the network with unlimited lifetime. To this end, the second part of the thesis focuses on the design and analysis of EH-aware MAC ...

Optimizing the performance of digital imaging systems with respect to the capture, display, storage and transmission of visual information represents one of the biggest challenges in the field of image and video processing. Taking into account the way humans perceive visual information can be greatly beneficial for this task. To achieve this, it is necessary to understand and model the human visual system, which is also the principal goal of this thesis. Computational models for different aspects of the visual system are developed, which can be used in a wide variety of image and video processing applications. The proposed models and metrics are shown to be consistent with human perception. The focus of this work is visual quality assessment. A perceptual distortion metric (PDM) for the evaluation of video quality is presented. It is based on a model of the ...

In order to meet ever-growing needs for capacity in wireless networks, transmission techniques and the system models used to study their performances have rapidly evolved. From single-user single-antenna point-to-point communications to modern multi-cell multi-antenna cellular networks there have been large advances in technology. Along the way, several assumptions are made in order to have either more realistic models, but also to allow simpler analysis. We analyze three aspects of actual networks and try to benefit from them when possible or conversely, to mitigate their negative impact. This sometimes corrects overly optimistic results, for instance when delay in the channel state information (CSI) acquisition is no longer neglected. However, this sometimes also corrects overly pessimistic results, for instance when in a broadcast channel (BC) the number of users is no longer limited to be equal to the number of transmit antennas ...

A large part of human speech communication takes place in noisy environments and is supported by technical devices. For example, a hearing-impaired person might use a hearing aid to take part in a conversation in a busy restaurant. These devices, but also telecommunication in noisy environments or voiced-controlled assistants, make use of speech enhancement and separation algorithms that improve the quality and intelligibility of speech by separating speakers and suppressing background noise as well as other unwanted effects such as reverberation. If the devices are equipped with more than one microphone, which is very common nowadays, then multi-channel speech enhancement approaches can leverage spatial information in addition to single-channel tempo-spectral information to perform the task. Traditionally, linear spatial filters, so-called beamformers, have been employed to suppress the signal components from other than the target direction and thereby enhance the desired ...

In digital speech-communication systems like mobile phones, public address systems and hearing aids, conveying the message is one of the most important goals. This can be challenging since the intelligibility of the speech may be harmed at various stages before, during and after the transmission process from sender to receiver. Causes which create such adverse conditions include background noise, an unreliable internet connection during a Skype conversation or a hearing impairment of the receiver. To overcome this, many speech-communication systems include speech processing algorithms to compensate for these signal degradations like noise reduction. To determine the effect on speech intelligibility of these signal processing based solutions, the speech signal has to be evaluated by means of a listening test with human listeners. However, such tests are costly and time consuming. As an alternative, reliable and fast machine-driven intelligibility predictors are ...

For the provisioning of video streaming services it is essential to provide a required level of customer satisfaction, given by the perceived video stream quality. It is therefore important to choose the compression parameters as well as the network settings so that they maximize the end-user quality. Due to video compression improvements of the newest video coding standard H.264/AVC, video streaming for low bit and frame rates is possible while preserving its perceptual quality. This is especially suitable for video applications in 3G wireless networks. Mobile video streaming is characterized by low resolutions and low bitrates. The commonly used resolutions are Quarter Common Intermediate Format (QCIF,176x144 pixels) for cell phones, Common Intermediate Format (CIF, 352x288 pixels) and Standard Interchange Format (SIF or QVGA, 320x240 pixels) for data-cards and palmtops (PDA). The mandatory codec for Universal Mobile Telecommunications System (UMTS) streaming ...

For a large class of non-delay-critical applications (e.g., buffered video streaming or data transfer from cloud services to local devices), end-to-end throughput becomes the most crucial key performance indicator (KPI). In cellular networks, the achievable end-user throughput (the maximum throughput a user will get when attempting to download as much data as possible) is a spatiotemporal function, and its estimation poses a challenging and as-yet unsolved problem. The ability to accurately predict achievable throughput in a given location and time interval would, for example, allow mobile operators to further optimize their networks and design more personalized offers for the customers, or allow end-users with mobile broadband modems to make more informed decisions when selecting a provider. This work investigates the impact of individual parameters on the end-user achievable throughput in cellular networks and analyzes the feasibility and limitations of constructing ...

THE task of modeling data traffic in networks is as old as the first commercial telephony systems. In the recent past in mobile telephone networks the focus has moved from voice to packetswitched services. The new cellular mobile networks of the third generation (UMTS) and the evolved second generation (GPRS) offer the subscriber the possibility of staying online everywhere and at any time. The design and dimensioning is well known for circuit switched voice systems, but not for mobile packet-switched systems. The terms user expectation, grade of service and so on need to be defined. To find these parameters it is important to have an accurate traffic model that delivers good traffic estimates. In this thesis we carried out measurements in a live 3G core network of an Austrian operator, in order to find appropriate models that can serve as ...

Nowadays, speech synthesis is part of various daily life applications. The ultimate goal of such technologies consists in extending the possibilities of interaction with the machine, in order to get closer to human-like communications. However, current state-of-the-art systems often lack of realism: although high-quality speech synthesis can be produced by many researchers and companies around the world, synthetic voices are generally perceived as hyperarticulated. In any case, their degree of articulation is fixed once and for all. The present thesis falls within the more general quest for enriching expressivity in speech synthesis. The main idea consists in improving statistical parametric speech synthesis, whose most famous example is Hidden Markov Model (HMM) based speech synthesis, by introducing a control of the articulation degree, so as to enable synthesizers to automatically adapt their way of speaking to the contextual situation, like humans ...