A zero-delay sequential scheme for lossy coding of individual sequences

We consider adaptive sequential lossy coding of bounded individual sequences when the performance is measured by the sequentially accumulated mean squared distortion. Theencoder and the decoder are connected via a noiseless channel of capacity $R$ and both are assumed to have zero delay. No probabilistic assumptions are made on how the sequence to be encoded is generated. For any bounded sequence of length $n$, the distortion redundancy is defined as the normalized cumulative distortion of the sequential scheme minus the normalized cumulative distortion of the best scalarquantizer of rate $R$ which is matched to this particular sequence. We demonstrate the existence of a zero-delay sequential scheme which uses common randomization in the encoder and the decoder such that the normalized maximum distortion redundancy converges to zero at a rate $n^{-1/5}\log n$ as the length of the encoded sequence $n$ increases without bound.

Augmented LDPC Graph for Distributed Video Coding with Multiple Side Information

The advances made in channel-capacity codes, such as turbo codes and low-density parity-check (LDPC) codes, have played a major role in the emerging distributed source coding paradigm. LDPC codes can be easily adapted to new source coding strategies due to their natural representation as bipartite graphs and the use of quasi-optimal decoding algorithms, such as belief propagation. This paper tackles a relevant scenario in distributedvideo coding: lossy source coding when multiple side information (SI) hypotheses are available at the decoder, each one correlated with the source according to different correlation noise channels. Thus, it is proposed to exploit multiple SI hypotheses through an efficient joint decoding technique withmultiple LDPC syndrome decoders that exchange information to obtain coding efficiency improvements. At the decoder side, the multiple SI hypotheses are created with motion compensated frame interpolation and fused together in a novel iterative LDPC based Slepian-Wolf decoding algorithm. With the creation of multiple SI hypotheses and the proposed decoding algorithm, bitrate savings up to 8.0% are obtained for similar decoded quality.

Perceptually driven video error protection using a distributed source coding approach

In video communication systems, the video signals are typically compressed and sent to the decoder through an error-prone transmission channel that may corrupt the compressed signal, causing the degradation of the final decoded video quality. In this context, it is possible to enhance the error resilience of typical predictive video coding schemes using as inspiration principles and tools from an alternative video coding approach, the so-called Distributed Video Coding (DVC), based on the Distributed Source Coding (DSC) theory. Further improvements in the decoded video quality after error-prone transmission may also be obtained by considering the perceptual relevance of the video content, as distortions occurring in different regions of a picture have a different impact on the user's final experience. In this context, this paper proposes a Perceptually Driven Error Protection (PDEP) video coding solution that enhances the error resilience of a state-of-the-art H.264/AVC predictive video codec using DSC principles and perceptual considerations. To increase the H.264/AVC error resilience performance, the main technical novelties brought by the proposed video coding solution are: (i) design of an improved compressed domain perceptual classification mechanism; (ii) design of an improved transcoding tool for the DSC-based protection mechanism; and (iii) integration of a perceptual classification mechanism in an H.264/AVC compliant codec with a DSC-based error protection mechanism. The performance results obtained show that the proposed PDEP video codec provides a better performing alternative to traditional error protection video coding schemes, notably Forward Error Correction (FEC)-based schemes. (C) 2013 Elsevier B.V. All rights reserved.

Assignment on Filters, Source Coding and Channel Coding

Considers bandpass filters, Huffman coding, arithmetic coding and Hamming coding.

Energy efficient transmission protocol for distributed source coding in sensor networks

Distributed source coding (DSC) has recently been considered as an efficient approach to data compression in wireless sensor networks (WSN). Using this coding method multiple sensor nodes compress their correlated observations without inter-node communications. Therefore energy and bandwidth can be efficiently saved. In this paper, we investigate a randombinning based DSC scheme for remote source estimation in WSN and its performance of estimated signal to distortion ratio (SDR). With the introduction of a detailed power consumption model for wireless sensor communications, we quantitatively analyze the overall network energy consumption of the DSC scheme. We further propose a novel energy-aware transmission protocol for the DSC scheme, which flexibly optimizes the DSC performance in terms of either SDR or energy consumption, by adapting the source coding and transmission parameters to the network conditions. Simulations validate the energy efficiency of the proposed adaptive transmission protocol. © 2007 IEEE.

Advanced side information creation techniques and framework for Wyner-Ziv video coding

Recently, several distributed video coding (DVC) solutions based on the distributed source coding (DSC) paradigm have appeared in the literature. Wyner-Ziv (WZ) video coding, a particular case of DVC where side information is made available at the decoder, enable to achieve a flexible distribution of the computational complexity between the encoder and decoder, promising to fulfill novel requirements from applications such as video surveillance, sensor networks and mobile camera phones. The quality of the side information at the decoder has a critical role in determining the WZ video coding rate-distortion (RD) performance, notably to raise it to a level as close as possible to the RD performance of standard predictive video coding schemes. Towards this target, efficient motion search algorithms for powerful frame interpolation are much needed at the decoder. In this paper, the RD performance of a Wyner-Ziv video codec is improved by using novel, advanced motion compensated frame interpolation techniques to generate the side information. The development of these type of side information estimators is a difficult problem in WZ video coding, especially because the decoder only has available some reference, decoded frames. Based on the regularization of the motion field, novel side information creation techniques are proposed in this paper along with a new frame interpolation framework able to generate higher quality side information at the decoder. To illustrate the RD performance improvements, this novel side information creation framework has been integrated in a transform domain turbo coding based Wyner-Ziv video codec. Experimental results show that the novel side information creation solution leads to better RD performance than available state-of-the-art side information estimators, with improvements up to 2 dB: moreover, it allows outperforming H.264/AVC Intra by up to 3 dB with a lower encoding complexity.

Source coded image data in the presence of channel errors

The growth and advances made in computer technology have led to the present interest in picture processing techniques. When considering image data compression the tendency is towards trans-form source coding of the image data. This method of source coding has reached a stage where very high reductions in the number of bits representing the data can be made while still preserving image fidelity. The point has thus been reached where channel errors need to be considered, as these will be inherent in any image comnunication system. The thesis first describes general source coding of images with the emphasis almost totally on transform coding. The transform technique adopted is the Discrete Cosine Transform (DCT) which becomes common to both transform coders. Hereafter the techniques of source coding differ substantially i.e. one tech­nique involves zonal coding, the other involves threshold coding. Having outlined the theory and methods of implementation of the two source coders, their performances are then assessed first in the absence, and then in the presence, of channel errors. These tests provide a foundation on which to base methods of protection against channel errors. Six different protection schemes are then proposed. Results obtained, from each particular, combined, source and channel error protection scheme, which are described in full are then presented. Comparisons are made between each scheme and indicate the best one to use given a particular channel error rate.

Distortion control for delay-sensitive sources

We investigate the problem of finding minimum-distortion policies for streaming delay-sensitive but distortion-tolerant data. We consider cross-layer approaches which exploit the coupling between presentation and transport layers. We make the natural assumption that the distortion function is convex and decreasing. We focus on a single source-destination pair and analytically find the optimum transmission policy when the transmission is done over an error-free channel. This optimum policy turns out to be independent of the exact form of the convex and decreasing distortion function. Then, for a packet-erasure channel, we analytically find the optimum open-loop transmission policy, which is also independent of the form of the convex distortion function. We then find computationally efficient closed-loop heuristic policies and show, through numerical evaluation, that they outperform the open-loop policy and have near optimal performance.

Interactive quantum information theory

La théorie de l'information quantique s'est développée à une vitesse fulgurante au cours des vingt dernières années, avec des analogues et extensions des théorèmes de codage de source et de codage sur canal bruité pour la communication unidirectionnelle. Pour la communication interactive, un analogue quantique de la complexité de la communication a été développé, pour lequel les protocoles quantiques peuvent performer exponentiellement mieux que les meilleurs protocoles classiques pour certaines tâches classiques. Cependant, l'information quantique est beaucoup plus sensible au bruit que l'information classique. Il est donc impératif d'utiliser les ressources quantiques à leur plein potentiel. Dans cette thèse, nous étudions les protocoles quantiques interactifs du point de vue de la théorie de l'information et étudions les analogues du codage de source et du codage sur canal bruité. Le cadre considéré est celui de la complexité de la communication: Alice et Bob veulent faire un calcul quantique biparti tout en minimisant la quantité de communication échangée, sans égard au coût des calculs locaux. Nos résultats sont séparés en trois chapitres distincts, qui sont organisés de sorte à ce que chacun puisse être lu indépendamment. Étant donné le rôle central qu'elle occupe dans le contexte de la compression interactive, un chapitre est dédié à l'étude de la tâche de la redistribution d'état quantique. Nous prouvons des bornes inférieures sur les coûts de communication nécessaires dans un contexte interactif. Nous prouvons également des bornes atteignables avec un seul message, dans un contexte d'usage unique. Dans un chapitre subséquent, nous définissons une nouvelle notion de complexité de l'information quantique. Celle-ci caractérise la quantité d'information, plutôt que de communication, qu'Alice et Bob doivent échanger pour calculer une tâche bipartie. Nous prouvons beaucoup de propriétés structurelles pour cette quantité, et nous lui donnons une interprétation opérationnelle en tant que complexité de la communication quantique amortie. Dans le cas particulier d'entrées classiques, nous donnons une autre caractérisation permettant de quantifier le coût encouru par un protocole quantique qui oublie de l'information classique. Deux applications sont présentées: le premier résultat général de somme directe pour la complexité de la communication quantique à plus d'une ronde, ainsi qu'une borne optimale, à un terme polylogarithmique près, pour la complexité de la communication quantique avec un nombre de rondes limité pour la fonction « ensembles disjoints ». Dans un chapitre final, nous initions l'étude de la capacité interactive quantique pour les canaux bruités. Étant donné que les techniques pour distribuer de l'intrication sont bien étudiées, nous nous concentrons sur un modèle avec intrication préalable parfaite et communication classique bruitée. Nous démontrons que dans le cadre plus ardu des erreurs adversarielles, nous pouvons tolérer un taux d'erreur maximal de une demie moins epsilon, avec epsilon plus grand que zéro arbitrairement petit, et ce avec un taux de communication positif. Il s'ensuit que les canaux avec bruit aléatoire ayant une capacité positive pour la transmission unidirectionnelle ont une capacité positive pour la communication interactive quantique. Nous concluons avec une discussion de nos résultats et des directions futures pour ce programme de recherche sur une théorie de l'information quantique interactive.

Quadratic Predictor based Differential Encoding and Decoding of Speech Signals

Modeling nonlinear systems using Volterra series is a century old method but practical realizations were hampered by inadequate hardware to handle the increased computational complexity stemming from its use. But interest is renewed recently, in designing and implementing filters which can model much of the polynomial nonlinearities inherent in practical systems. The key advantage in resorting to Volterra power series for this purpose is that nonlinear filters so designed can be made to work in parallel with the existing LTI systems, yielding improved performance. This paper describes the inclusion of a quadratic predictor (with nonlinearity order 2) with a linear predictor in an analog source coding system. Analog coding schemes generally ignore the source generation mechanisms but focuses on high fidelity reconstruction at the receiver. The widely used method of differential pnlse code modulation (DPCM) for speech transmission uses a linear predictor to estimate the next possible value of the input speech signal. But this linear system do not account for the inherent nonlinearities in speech signals arising out of multiple reflections in the vocal tract. So a quadratic predictor is designed and implemented in parallel with the linear predictor to yield improved mean square error performance. The augmented speech coder is tested on speech signals transmitted over an additive white gaussian noise (AWGN) channel.

Codificação de fonte

Pós-graduação em Matemática Universitária - IGCE

Adaptive wireless multimedia communication systems

In recent years, due to the rapid convergence of multimedia services, Internet and wireless communications, there has been a growing trend of heterogeneity (in terms of channel bandwidths, mobility levels of terminals, end-user quality-of-service (QoS) requirements) for emerging integrated wired/wireless networks. Moreover, in nowadays systems, a multitude of users coexists within the same network, each of them with his own QoS requirement and bandwidth availability. In this framework, embedded source coding allowing partial decoding at various resolution is an appealing technique for multimedia transmissions. This dissertation includes my PhD research, mainly devoted to the study of embedded multimedia bitstreams in heterogenous networks, developed at the University of Bologna, advised by Prof. O. Andrisano and Prof. A. Conti, and at the University of California, San Diego (UCSD), where I spent eighteen months as a visiting scholar, advised by Prof. L. B. Milstein and Prof. P. C. Cosman. In order to improve the multimedia transmission quality over wireless channels, joint source and channel coding optimization is investigated in a 2D time-frequency resource block for an OFDM system. We show that knowing the order of diversity in time and/or frequency domain can assist image (video) coding in selecting optimal channel code rates (source and channel code rates). Then, adaptive modulation techniques, aimed at maximizing the spectral efficiency, are investigated as another possible solution for improving multimedia transmissions. For both slow and fast adaptive modulations, the effects of imperfect channel estimation errors are evaluated, showing that the fast technique, optimal in ideal systems, might be outperformed by the slow adaptive modulation, when a real test case is considered. Finally, the effects of co-channel interference and approximated bit error probability (BEP) are evaluated in adaptive modulation techniques, providing new decision regions concepts, and showing how the widely used BEP approximations lead to a substantial loss in the overall performance.

Estructura y pruebas con Labmu

La implantación de la televisión digital en España ha supuesto un conjunto de desafíos técnicos y de orden práctico que se han ido acometiendo en multitud de ámbitos, desde la legislación que normaliza las infraestructuras comunes de telecomunicación hasta los cambios en las instalaciones y receptores donde el usuario final recibe los servicios. Por la complejidad y el carácter interdisciplinar de los conocimientos necesarios el aprendizaje de los titulados dentro del ámbito de la Telecomunicación supone también un reto importante. Este proyecto realiza una primera aproximación a un conjunto de herramientas hardware y software de ayuda a la enseñanza de esta amplia disciplina. El proyecto se ha realizado en torno a Labmu laboratorio multiusuario para prácticas de televisión digital de la empresa Xpertia. Se ha realizado para conocer y documentar sus posibilidades. También se ha documentado la tarjeta moduladora DTA-111 y el software para Windows StreamXpress. Estos sistemas ofrecen muchas posibilidades para la docencia de la televisión digital en todas las áreas desde la codificación fuente hasta la decodificación en el usuario final. En particular para ambos sistemas se han realizado pruebas en radiofrecuencia de emisiones de TDT. También se han establecido algunas ideas para trabajo futuro con estos sistemas. El proyecto se divide en seis capítulos: Capitulo 1: En el primer capítulo titulado Introducción se presenta el proyecto. Capítulo 2: En el segundo capítulo titulado Composición Hardware Labmu se presentan todos los componentes del laboratorio Labmu con la descripción de cada componente y sus características técnicas. Así mismo se presenta el interconexionado y configuración con que se ha trabajado. Capítulo 3: En el tercer capítulo titulado Software Labmu se describe como manual de usuario todos los componentes y posibilidades de software de Labmu. Capítulo 4: En el cuarto capítulo titulado Medidas con Labmu se realiza las medidas de MER, CBER, VBER, C/N y potencia de canal de los canales emitidos en la Comunidad de Madrid, comparando estas medidas con el analizador Promax Prodig-5. Capítulo 5: En el quinto capítulo titulado Tarjeta receptora y software se describe la tarjeta DTA-111 y el software StreamXpress, realizando medidas con el analizador Promax Prodig-5 introduciendo errores a la señal emitida por la tarjeta, y estudiando los niveles límites de visualización correcta. Capítulo 6: En el sexto y último capítulo titulado Conclusiones se presentan las conclusiones del proyecto y un plan de trabajo futuro ABSTRACT. The implantation of Digital Television in Spain (TDT) has implied a number of technical and practical challenges in several scopes. These challenges range from recommendations that standardize common telecommunications infrastructure to the changes in facilities where the end user receives digital services. The complexity and the interdisciplinary nature of skills that graduate Telecommunications students need to learn, is also a major challenge. This project is a first approach of a set of hard ware and software tools to help in the task of teaching this broad range discipline. The project has been carried out on Labmu, a multiuser Digital Television laboratory created by the Xpertia company. Its objectives are to understand and document this range of possibilities. DTA-111 modulator card and software for Windows StreamXpress have also been documented. These systems offer many options for teaching digital television in all areas, from source coding to end user decoding. In particular, both systems were tested on RF emissions in TDT. More over some ideas for future work with these systems have also foreseen. The project is structured in six chapters: Chapter1: This section Introduces the project. Chapter2: Titled “The Composition Hardware Labmu” presents Labmu lab components and provides descriptions for each component and its technical characteristics. It also presents the interconnection and configuration we have been using. Chapter3: Titled “Software Labmu” is a user manual, describing all components and software possibilities Labmu offers. Chapter4: Titled “Measures to Labmu” presents MER, CBER, VBER, C /N and channel power measures provided by Labmu in comparison with Promax Prodig-5 measures for all channels broad casting digital television services in the Community of Madrid. Chapter 5 Titled “Receiver card and software” describes DTA-111 card and software StreamXpress. Also the effects of errors insertion performed by this card are measured with the PromaxProdig-5 meter. Threes hold levels for correct reception are also studied. Chapter6: Entitled Conclusions presents the conclusions of the project and a plan for future work.

Improved detection in CDMA for biased sources

We consider the detection of biased information sources in the ubiquitous code-division multiple-access (CDMA) scheme. We propose a simple modification to both the popular single-user matched-filter detector and a recently introduced near-optimal message-passing-based multiuser detector. This modification allows for detecting modulated biased sources directly with no need for source coding. Analytical results and simulations with excellent agreement are provided, demonstrating substantial improvement in bit error rate in comparison with the unmodified detectors and the alternative of source compression. The robustness of error-performance improvement is shown under practical model settings, including bias estimation mismatch and finite-length spreading codes. © 2007 IOP Publishing Ltd.

DNA bar-coding reveals source and patterns of Thaumastocoris peregrinus invasions in South Africa and South America

Thaumastocoris peregrinus is a recently introduced invertebrate pest of non-native Eucalyptus plantations in the Southern Hemisphere. It was first reported from South Africa in 2003 and in Argentina in 2005. Since then, populations have grown explosively and it has attained an almost ubiquitous distribution over several regions in South Africa on 26 Eucalyptus species. Here we address three key questions regarding this invasion, namely whether only one species has been introduced, whether there were single or multiple introductions into South Africa and South America and what the source of the introduction might have been. To answer these questions, bar-coding using mitochondrial DNA (COI) sequence diversity was used to characterise the populations of this insect from Australia, Argentina, Brazil, South Africa and Uruguay. Analyses revealed three cryptic species in Australia, of which only T. peregrinus is represented in South Africa and South America. Thaumastocoris peregrinus populations contained eight haplotypes, with a pairwise nucleotide distance of 0.2-0.9% from seventeen locations in Australia. Three of these haplotypes are shared with populations in South America and South Africa, but the latter regions do not share haplotypes. These data, together with the current distribution of the haplotypes and the known direction of original spread in these regions, suggest that at least three distinct introductions of the insect occurred in South Africa and South America before 2005. The two most common haplotypes in Sydney, one of which was also found in Brisbane, are shared with the non-native regions. Sydney populations of T. peregrinus, which have regularly reached outbreak levels in recent years, might thus have served as source of these three distinct introductions into other regions of the Southern Hemisphere.