Cross-Layer Resource Allocation Protocols for Multimedia CDMA Networks

The design of mechanisms to efficiently allow many users to maintain simultaneous communications while sharing the same transmission medium is a crucial step during a wireless network design. The resource allocation process needs to meet numerous requirements that are sometimes conflicting, such as high efficiency, network utilization and flexibility and good communication quality. Due to limited resources, wireless cellular networks are normally seen as having some limit on the network capacity, in terms of the maximum number of calls that may be supported. Being able to dynamically extend network operation beyond the set limit at the cost of a smooth and small increase in distortion is a valuable and useful idea because it provides the means to flexibly adjust the network to situations where it is more important to service a call rather than to guarantee the best quality. In this thesis we study designs for resource allocation in CDMA networks carrying conversational-type calls. The designs are based on a cross-layer approach where the source encoder, the channel encoder and, in some cases, the processing gains are adapted. The primary focus of the study is on optimally multiplexing multimedia sources. Therefore, we study optimal resource allocation to resolve interference-generated congestion for an arbitrary set of real-time variable-rate source encoders in a multimedia CDMA network. Importantly, we show that the problem could be viewed as the one of statistical multiplexing in source-adapted multimedia CDMA. We present analysis and optimal solutions for different system setups. The result is a flexible system that sets an efficient tradeoff between end-to-end distortion and number of users. Because in the presented cross-layer designs channel-induced errors are kept at a subjectively acceptable level, the proposed designs are able to outperform equivalent CDMA systems where capacity is increased in the traditional way, by allowing a reduction in SINR. An important application and part of this study, is the use of the proposed designs to extend operation of the CDMA network beyond a defined congestion operating point. Also, the general framework for statistical multiplexing in CDMA is used to study some issues in integrated real-time/data networks

A General Framework for Analyzing, Characterizing, and Implementing Spectrally Modulated, Spectrally Encoded Signals

Fourth generation (4G) communications will support many capabilities while providing universal, high speed access. One potential enabler for these capabilities is software defined radio (SDR). When controlled by cognitive radio (CR) principles, the required waveform diversity is achieved via a synergistic union called CR-based SDR. Research is rapidly progressing in SDR hardware and software venues, but current CR-based SDR research lacks the theoretical foundation and analytic framework to permit efficient implementation. This limitation is addressed here by introducing a general framework for analyzing, characterizing, and implementing spectrally modulated, spectrally encoded (SMSE) signals within CR-based SDR architectures. Given orthogonal frequency division multiplexing (OFDM) is a 4G candidate signal, OFDM-based signals are collectively classified as SMSE since modulation and encoding are spectrally applied. The proposed framework provides analytic commonality and unification of SMSE signals. Applicability is first shown for candidate 4G signals, and resultant analytic expressions agree with published results. Implementability is then demonstrated in multiple coexistence scenarios via modeling and simulation to reinforce practical utility

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook

Study of the cellular mobile communication: services perspective comparisons and implementation analysis of 3G services against two 3G standards

EEE/201

Perceptual and acoustic impacts of aberrant properties of electrolaryngeal speech.

Thesis (Ph. D.)—Harvard-MIT Division of Health Sciences and Technology, 2003.Includes bibliographical references (p. 167-171).This electronic version was prepared by the author. The certified thesis is available in the Institute Archives and Special Collections.Ph. D

Détection et modification des transitoires d'un signal de parole dans le but de rendre un codec plus robuste aux pertes de paquets

Pour transmettre les signaux de parole de façon efficace, ces derniers sont compressés et transmis en trames typiquement de 10 à 20 ms. Lors de la transmission des trames, il arrive que ces dernières soient perdues. Lors de la reconstruction du signal au décodeur, il est préférable de remplacer les trames perdues par un signal qui se rapproche le plus possible du signal manquant. Le signal perdu est souvent reconstruit en se basant sur l'information des dernières trames reçues, puisque, de façon générale, les propriétés statistiques du signal de parole évoluent relativement lentement d'une trame à la suivante. Les signaux de parole peuvent être classés en différentes catégories (parole voisée, nonvoisée, transitoire, etc.). Afin de mieux exploiter les caractéristiques de chaque catégorie, il est pertinent d'appliquer un classificateur à chaque trame de signal. Cette classification des signaux permet un meilleur camouflage des trames perdues, optimisé pour les différentes classes. La classification des trames est parfois imprécise lors des transitions entre une trame non-voisée et une trame voisée. Ces erreurs de classification entraînent de mauvaises reconstructions de signal lors des pertes de trames. Pour pallier ces erreurs, cette Thèse propose un nouvel algorithme robuste qui identifie les trames critiques et qui applique la classification appropriée. Pour les trames dont les propriétés ne correspondent pas exactement à l'une des classes disponibles, une modification transparente du signal est appliquée pour rendre ces trames conformes à la classification proposée. Ces modifications permettent d'obtenir une meilleure reconstruction du signal si les trames suivantes sont perdues