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The thesis is concerned with the estimation of the sampled impulse-response of time-varying voiceband channels, and in particular with the proposed synchronous serial transmission of 16-level quadrature amplitude modulated digital data signals at 9600 bit/s over HF radio links. With such a system, the optimum detector at the receiver is a maximum likelihood detector implemented, for example, using the Viterbi algorithm. In this case, the detector requires knowledge of the sampled impulse-response of the channel. Channel estimators can also be used for estimating the response of any time-varying linear bandpass channel and need not be restricted in use only with a maximum likelihood detector. They may be employed in any such application where a time-varying channel must be tracked to ensure the correct operation of the detector. The thesis includes a description of the ionospheric propagation medium, with particular emphasis on the nature of the impairments that are likely to be encountered by the data signal. An appropriate model of the HF channel is simulated for subsequent use in testing the channel estimators. A summary is also given of the more important forms of channel estimators that are used for time-invariant or slowly time-varying channels. The characteristics of the HF radio medium may vary rapidly with time, so an estimator based on the Kalman filter is investigated in order to exploit the fast tracking capability of the filter. It is shown that inadequate modelling of the channel by the Kalman filter results in suboptimum performance (in the minimum mean square error sense) of the estimator, however, this can be improved by including a suitable predictor. The performance of the Kalman filter estimator, with and without the predictor, is then compared with the corresponding estimator which uses a feedforward transversal filter. The recently developed HF channel estimator based on the feedforward transversal-filter estimator is also investigated, but it is tested here over the simulated HF radio links with three independent Rayleigh fading sky waves, which represent typical poor conditions over actual links. Various degrees of prediction are also studied and based on the results, a change in the degree of prediction from that previously proposed is suggested as a better arrangement for use with the estimator when there are three sky waves. Finally, it is shown that a considerable reduction in the equipment complexity can be achieved by exploiting a selfcorrecting property of the estimator that has been discovered.

With the existing telephone networks evolving towards Integrated Services Digital Networks, there is a specific need to transmit high quality wideband (0-7 kHz) speech at 64 kbps or below for special services like voice channels in teleconferencing, commentary channels for broadcasting etc. In this thesis, a computer simulation study of digital coding of wideband speech at 64 kbps using relatively simple coding techniques is first presented. The performance of ADPCM coders employing fixed or adaptive prediction, with or without noise spectral shaping, and 2-band subband coders is examined under ideal as well as noisy channel conditions. While preserving the quality of the 64 kbps recovered speech, the transmission bit rate is reduced to 56 kbps so that 8 kbps data can be accommodated within 64 kbps channel.

An underwater acoustic voice communications system can provide a vital communication link between divers and surface supervisors. There are numerous situations in which a communication system is essential. In the event of an emergency, a diver's life may depend on fast and effective action at the surface. The design and implementation of a digital underwater acoustic voice communication system using a digital signal processor (DSP) is described. The use of a DSP enables the adoption of computationally complex speech signal processing algorithms and the transmission and reception of digital data through an underwater acoustic channel. The system is capable of operating in both transmitting and receiving modes by using a mode selection scheme. During the transmission mode, by using linear predictive coding (LPC), the speech signal is compressed whilst transmitting the compressed data in digital pulse position modulation (DPPM) format at a transmission rate of 2400 bps. At the receiver, a maximum energy detection technique is employed to identify the pulse position, enabling correct data decoding which in turn allows the speech signal to be reconstructed. The advantage of the system is to introduce advances in digital technology to underwater acoustic voice communications and update the present analogue systems employing AM and SSB modulation. Since the DSP-based system is designed in modular sections, the hardware and software can be modified if the performance of the system is inadequate. The communication system was tested successfully in a large indoor tank to simulate the effect of a short and very shallow underwater channel with severe multipath reverberation. The other objective of this study was to improve the quality of the transmitted speech signal. When the system is used by SCUBA divers, the speech signal is produced in a mask with a high pressure air environment, and bubble and breathing noise affect the speech clarity. Breathing noise is cancelled by implementing a combination of zero crossing rate and energy detection. In order to cancel bubble noise spectral subtraction and adaptive noise cancelling algorithms were simulated; the latter was found to be superior and was adopted for the current system.

There is a lot of interest at present in the provision of an integrated network which can carry both voice and data traffic. The recent emergence of asynchronous transfer mode (ATM) networks has made this desire a reality. The Unison project built a network consisting of local area networks (LANs) and exchange LANs connected at local and remote sites by an ATM-Iike backbone network. This extends the facilities of the LAN to a wider area. The Unison network is also an intelligent network, because it provides a service to the user (such as dynamic data-bases, which ease call set-up and facilitate user migration). An important feature of the Unison network is also the provision of applications which demonstrate the suitability of the network to carry both voice and data traffic, and which exploit the intelligent network features. This thesis describes a very important application: the provision of a two- and three-party wide-band speech teleconferencing system. The first part of the thesis deals with the provision of a two-party teleconferencing system, based on a wide-band speech codec. The codec is interfaced to the Unison Network via transputers (parallel processors). This thesis considers in detail the voice protocols which make up part of the network interface. The work includes the set-up and control benefits gained from interaction with a desk-top workstation, which can also be used to guide other multi-media services (such as video). A topic which has been greatly under-stressed in other similar research (i.e. the acoustic aspects of the system) is also investigated. The second part of the thesis deals with the logical expansion of the two-party system to a three- or more party scenario across the Unison network. Towards this end, a bridge has been designed and implemented based on transputers. The problems associated with matching the DSP algorithm used in the codecs with that implemented in the bridge is also discussed. The same systems considerations addressed by the two-party version are expanded to operate with the three-party teleconferencing system.

The thesis is concerned with the design of a phase-shift keying system for a digital modem, operating over a satellite link. Computer simulation tests and theoretical analyses are used to assess the proposed design. The optimum design of both transmitter and receiver filters for the system to be used in the modem are discussed. Sinusoidal roll-off spectrum with different roll-off factor and optimum truncation lengths of the sample impulse response are designed for the proposed scheme to approximate to the theoretical ideal. It has used an EF bandpass filter to band limit the modulated signal, which forms part of the satellite channel modelling. The high power amplifier (HPA) at the earth station has been used in the satellite channel modelling due to its effect in introducing nonlinear AMAM and AM-PM conversion effects and distortion on the transmitted signal from the earth station. The satellite transponder is assumed to be operating in a linear mode. Different phase-shift keying signals such as differentially encoded quaternary phase-shift keying (DEQPSK), offset quaternary phase-shift keying (OQPSK) and convolutionally encoded 8PSK (CE8PSK) signals are analysed and discussed in the thesis, when the high power amplifier (HPA) at the earth station is operating in a nonlinear mode. Convolutional encoding is discussed when applied to the system used in the modem, and a Viterbi -algorithm decoder at the receiver has been used, for CE8PSK signals for a nonlinear satellite channel. A method of feed-forward synchronisation scheme is designed for carrier recovery in CE8PSK receiver. The thesis describes a method of baseband linearizing the baseband signal in order to reduce the nonlinear effects caused by the HPA at the earth station. The scheme which compensates for the nonlinear effects of the HPA by predistorting the baseband signal prior to modulation as opposed to correcting the distortion after modulation, thus reducing the effects of nonlinear distortion introduced by the HPA. The results of the improvement are presented. The advanced technology of digital signal processors (DSPs) has been used in the implementation of the demodulation and digital filtering parts of the modem replacing large parts of conventional circuits. The Viterbi-algorithm decoder for CE8PSK signals has been implemented using a digital signal processor chip, giving excellent performance and is a cost effective and easy way for future developments and any modifications, The results showed that, by using the various studied techniques, as well as the implementation of digital signal processor chip in parts of the modem, a potentially more cost effective modem can be obtained.

This thesis describes the application of VLSI devices to channel evaluation and communication techniques over ionospheric radio paths. Digital signal processing techniques using microprocessors and charge coupled devices are described in detail. A novel method for observing interference and fading patterns on HF channels is described. Error control coding schemes and digital modulation techniques are combined in a design for an adaptive modem for use over HF radio links. Results of narrow-band interference measurements, error patterns and coding performance are presented.

This thesis investigates, using simulation techniques, the practical aspects of implementing a novel mobility protocol on the emerging Broadband Integrated Services Digital Network standard. The increasing expansion of telecommunications networks has meant that the demand for simulation has increased rapidly in recent years; but conventional simulators are slow and developments in the communications field are outstripping the ability of sequential uni-processor simulators. Newer techniques using distributed simulation on a multi-processor network are investigated in an attempt to make a cell-level simulation of a non-trivial B.-I.S.D.N. network feasible. The current state of development of the Asynchronous Transfer Mode standard, which will be used to implement a B.-I.S.D.N., is reviewed and simulation studies of the Orwell Slotted Ring protocol were made in an attempt to devise a simpler model for use in the main simulator. The mobility protocol, which uses a footprinting technique to simplify hand- offs by distributing information about a connexion to surrounding base stations, was implemented on the simulator and found to be functional after a few 'special case' scenarios had been catered for.

No description available.

No description available.