NASA's mobile satellite development program

A Mobile Satellite System (MSS) will provide data and voice communications over a vast geographical area to a large population of mobile users. A technical overview is given of the extensive research and development studies and development performed under NASA's mobile satellite program (MSAT-X) in support of the introduction of a U.S. MSS. The critical technologies necessary to enable such a system are emphasized: vehicle antennas, modulation and coding, speech coders, networking and propagation characterization. Also proposed is a first, and future generation MSS architecture based upon realized ground segment equipment and advanced space segment studies

MSAT-X: A technical introduction and status report

A technical introduction and status report for the Mobile Satellite Experiment (MSAT-X) program is presented. The concepts of a Mobile Satellite System (MSS) and its unique challenges are introduced. MSAT-X's role and objectives are delineated with focus on its achievements. An outline of MSS design philosophy is followed by a presentation and analysis of the MSAT-X results, which are cast in a broader context of an MSS. The current phase of MSAT-X has focused notably on the ground segment of MSS. The accomplishments in the four critical technology areas of vehicle antennas, modem and mobile terminal design, speech coding, and networking are presented. A concise evolutionary trace is incorporated in each area to elucidate the rationale leading to the current design choices. The findings in the area of propagation channel modeling are also summarized and their impact on system design discussed. To facilitate the assessment of the MSAT-X results, technology and subsystem recommendations are also included and integrated with a quantitative first-generation MSS design

Secure mobile radio communication over narrowband RF channel.

by Wong Chun Kau, Jolly.Thesis (M.Phil.)--Chinese University of Hong Kong, 1992.Includes bibliographical references (leaves 84-88).ABSTRACT --- p.1ACKNOWLEDGEMENT --- p.3Chapter 1. --- INTRODUCTION --- p.7Chapter 1.1 --- Land Mobile Radio (LMR) CommunicationsChapter 1.2 --- Paramilitary Communications SecurityChapter 1.3 --- Voice Scrambling MethodsChapter 1.4 --- Digital Voice EncryptionChapter 1.5 --- Digital Secure LMRChapter 2. --- DESIGN GOALS --- p.20Chapter 2.1 --- System Concept and ConfigurationChapter 2.2 --- Operational RequirementsChapter 2.2.1 --- Operating conditionsChapter 2.2.2 --- Intelligibility and speech qualityChapter 2.2.3 --- Field coverage and transmission delayChapter 2.2.4 --- Reliability and maintenanceChapter 2.3 --- Functional RequirementsChapter 2.3.1 --- Major system featuresChapter 2.3.2 --- Cryptographic featuresChapter 2.3.3 --- Phone patch facilityChapter 2.3.4 --- Mobile data capabilityChapter 2.4 --- Bandwidth RequirementsChapter 2.5 --- Bit Error Rate RequirementsChapter 3. --- VOICE CODERS --- p.38Chapter 3.1 --- Digital Speech Coding MethodsChapter 3.1.1 --- Waveform codingChapter 3.1.2 --- Linear predictive codingChapter 3.1.3 --- Sub-band codingChapter 3.1.4 --- VocodersChapter 3.2 --- Performance EvaluationChapter 4. --- CRYPTOGRAPHIC CONCERNS --- p.52Chapter 4.1 --- Basic Concepts and CryptoanalysisChapter 4.2 --- Digital Encryption TechniquesChapter 4.3 --- Crypto SynchronizationChapter 4.3.1 --- Auto synchronizationChapter 4.3.2 --- Initial synchronizationChapter 4.3.3 --- Continuous synchronizationChapter 4.3.4 --- Hybrid synchronizationChapter 5. --- DIGITAL MODULATION --- p.63Chapter 5.1 --- Narrowband Channel RequirementsChapter 5.2 --- Narrowband Digital FMChapter 5.3 --- Performance EvaluationChapter 6. --- SYSTEM IMPLEMENTATION --- p.71Chapter 6.1 --- Potential EMC ProblemsChapter 6.2 --- Frequency PlanningChapter 6.3 --- Key ManagementChapter 6.4 --- Potential Electromagnetic Compatibility (EMC) ProblemsChapter 7. --- CONCLUSION --- p.80LIST OF ILLUSTRATIONS --- p.81REFERENCES --- p.82APPENDICES --- p.89Chapter I. --- Path Propagation Loss(L) Vs Distance (d)Chapter II. --- Speech Quality Assessment Tests performedby Special Duties Unit (SDU

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

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

Proceedings of the Mobile Satellite Conference

A satellite-based mobile communications system provides voice and data communications to mobile users over a vast geographic area. The technical and service characteristics of mobile satellite systems (MSSs) are presented and form an in-depth view of the current MSS status at the system and subsystem levels. Major emphasis is placed on developments, current and future, in the following critical MSS technology areas: vehicle antennas, networking, modulation and coding, speech compression, channel characterization, space segment technology and MSS experiments. Also, the mobile satellite communications needs of government agencies are addressed, as is the MSS potential to fulfill them

Quantisation mechanisms in multi-protoype waveform coding

Prototype Waveform Coding is one of the most promising methods for speech coding at low bit rates over telecommunications networks. This thesis investigates quantisation mechanisms in Multi-Prototype Waveform (MPW) coding, and two prototype waveform quantisation algorithms for speech coding at bit rates of 2.4kb/s are proposed. Speech coders based on these algorithms have been found to be capable of producing coded speech with equivalent perceptual quality to that generated by the US 1016 Federal Standard CELP-4.8kb/s algorithm. The two proposed prototype waveform quantisation algorithms are based on Prototype Waveform Interpolation (PWI). The first algorithm is in an open loop architecture (Open Loop Quantisation). In this algorithm, the speech residual is represented as a series of prototype waveforms (PWs). The PWs are extracted in both voiced and unvoiced speech, time aligned and quantised and, at the receiver, the excitation is reconstructed by smooth interpolation between them. For low bit rate coding, the PW is decomposed into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW). The SEW is coded using vector quantisation on both magnitude and phase spectra. The SEW codebook search is based on the best matching of the SEW and the SEW codebook vector. The REW phase spectra is not quantised, but it is recovered using Gaussian noise. The REW magnitude spectra, on the other hand, can be either quantised with a certain update rate or only derived according to SEW behaviours

Air interface standards for digital mobile cellular systems in the U.S., Europe and Japan

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1995.Includes bibliographical references (leaves 152-156).by Rohit Sakhuja.M.Eng