TDRSS telecommunications system, PN code analysis

The pseudo noise (PN) codes required to support the TDRSS telecommunications services are analyzed and the impact of alternate coding techniques on the user transponder equipment, the TDRSS equipment, and all factors that contribute to the acquisition and performance of these telecommunication services is assessed. Possible alternatives to the currently proposed hybrid FH/direct sequence acquisition procedures are considered and compared relative to acquisition time, implementation complexity, operational reliability, and cost. The hybrid FH/direct sequence technique is analyzed and rejected in favor of a recommended approach which minimizes acquisition time and user transponder complexity while maximizing probability of acquisition and overall link reliability

Application of bit-slice microprocessors to digital correlation in spread spectrum communication systems

This thesis describes the application of commercially available microprocessors and other VLSI devices to high-speed real-time digital correlation in spread spectrum and related communication applications. Spread spectrum communications are a wide-band secure communication system that generate a very broad spectral bandwidth signal that is therefore hard to detect in noise. They are capable of rejecting intentional or unintentional jamming, and are insensitive to the multipath and fading that affects conventional high frequency systems. The bandwidth of spread spectrum systems must be large to obtain a significant performance improvement. This means that the sequence rate must be fast and therefore very fast microprocessors will be required when they are used to perform spread spectrum correlation. Since multiplication cannot be performed efficiently by microprocessors considerable work, since 1974, has been published in the literature which is devoted to minimising the requirement of multiplications in digital correlation and other signal processing algorithms. These fast techniques are investigated and implemented using general-purpose microprocessors. The restricted-bandwidth problem in microprocessor-based digital correlator has been discussed. A new implementation is suggested which uses bit-slice devices to maintain the flexibility of microprocessor-based digital correlation without sacrificing speed. This microprocessor-based system has been found to be efficient in implementing the correlation process at the baseband in the digital domain as well as the post-correlation signal processing- demodulation, detection and tracking, especiaJIy for low rate signals. A charge coupled-device is used to obtain spectral density function. An all-digital technique which is programmable for any binary waveform and can be used for achieving initial acquisition and maintaining synchronisation in spread spectrum communications is described. Many of the practical implementation problems are discussed. The receiver performance, which is measured in terms of the acquisition time and the bit-error rate, is also presented and results are obtained which are close to those predicted in the system simulations

TDRSS telecommunications study. Phase 1: Final report

A parametric analysis of the telecommunications support capability of the Tracking and Data Relay Satellite System (TDRSS) was performed. Emphasis was placed on maximizing support capability provided to the user while minimizing impact on the user spacecraft. This study evaluates the present TDRSS configuration as presented in the TDRSS Definition Phase Study Report, December 1973 to determine potential changes for improving the overall performance. In addition, it provides specifications of the user transponder equipment to be used in the TDRSS

Direct sequence spread spectrum techniques in local area networks

This thesis describes the application of a direct sequence spread spectrum modulation scheme to the physical layer of a local area networks subsequently named the SS-LAN. Most present day LANs employ erne form or another of time division multiplexing which performs well in many systems but which is limited by its very nature in real time, time critical and time demanding applications. The use of spread spectrum multiplexing removes these limitations by providing a simultaneous multiple user access capability to the channel which permits each and all nodes to utilise the channel independent of the activity being currently supported by that channel. The theory of spectral spreading is a consequence of the Shannon channel capacity in which the channel capacity may be maintained by the trading of signal to noise ratio for bandwidth. The increased bandwidth provides an increased signal dimensionality which can be utilised in providing noise immunity and/or a simultaneous multiple user environment: the effects of the simultaneous users can be considered as noise from the point of view of any particular constituent signal. The use of code sequences at the physical layer of a LAN permits a wide range of mapping alternatives which can be selected according to the particular application. Each of the mapping techniques possess the general spread spectrum properties but certain properties can be emphasised at the expense of others. The work has Involved the description of the properties of the SS-LAN coupled with the development of the mapping techniques for use In the distribution of the code sequences. This has been followed by an appraisal of a set of code sequences which has resulted in the definition of the ideal code properties and the selection of code families for particular types of applications. The top level design specification for the hardware required in the construction of the SS-LAN has also been presented and this has provided the basis for a simplified and idealised theoretical analysis of the performance parameters of the SS-LAN. A positive set of conclusions for the range of these parameters has been obtained and these have been further analysed by the use of a SS-LAN computer simulation program. This program can simulate any configuration of the SS-LAN and the results it has produced have been compared with those of the analysis and have been found to be in agreement. A tool for the further analysis of complex SS-LAN configurations has therefore been developed and this will form the basis for further work

Applications of the orthogonal property of binary chain codes

Imperial Users onl

Multipath/modulation study for the tracking and data relay satellite system Final report, 14 Apr. 1969 - 12 Jan. 1970

Multipath modulation study of tracking and data relay satellite syste

Differential encoding techniques applied to speech signals

The increasing use of digital communication systems has produced a continuous search for efficient methods of speech encoding. This thesis describes investigations of novel differential encoding systems. Initially Linear First Order DPCM systems employing a simple delayed encoding algorithm are examined. The systems detect an overload condition in the encoder, and through a simple algorithm reduce the overload noise at the expense of some increase in the quantization (granular) noise. The signal-to-noise ratio (snr) performance of such d codec has 1 to 2 dB's advantage compared to the First Order Linear DPCM system. In order to obtain a large improvement in snr the high correlation between successive pitch periods as well as the correlation between successive samples in the voiced speech waveform is exploited. A system called "Pitch Synchronous First Order DPCM" (PSFOD) has been developed. Here the difference Sequence formed between the samples of the input sequence in the current pitch period and the samples of the stored decoded sequence from the previous pitch period are encoded. This difference sequence has a smaller dynamic range than the original input speech sequence enabling a quantizer with better resolution to be used for the same transmission bit rate. The snr is increased by 6 dB compared with the peak snr of a First Order DPCM codea. A development of the PSFOD system called a Pitch Synchronous Differential Predictive Encoding system (PSDPE) is next investigated. The principle of its operation is to predict the next sample in the voiced-speech waveform, and form the prediction error which is then subtracted from the corresponding decoded prediction error in the previous pitch period. The difference is then encoded and transmitted. The improvement in snr is approximately 8 dB compared to an ADPCM codea, when the PSDPE system uses an adaptive PCM encoder. The snr of the system increases further when the efficiency of the predictors used improve. However, the performance of a predictor in any differential system is closely related to the quantizer used. The better the quantization the more information is available to the predictor and the better the prediction of the incoming speech samples. This leads automatically to the investigation in techniques of efficient quantization. A novel adaptive quantization technique called Dynamic Ratio quantizer (DRQ) is then considered and its theory presented. The quantizer uses an adaptive non-linear element which transforms the input samples of any amplitude to samples within a defined amplitude range. A fixed uniform quantizer quantizes the transformed signal. The snr for this quantizer is almost constant over a range of input power limited in practice by the dynamia range of the adaptive non-linear element, and it is 2 to 3 dB's better than the snr of a One Word Memory adaptive quantizer. Digital computer simulation techniques have been used widely in the above investigations and provide the necessary experimental flexibility. Their use is described in the text