Discrete-time audio equalisers introduce a variety of undesirable artefacts into audio mixing systems, namely, distortions caused by finite wordlength constraints, frequency response distortion due to coefficient calculation and signal disturbances that arise from real-time coefficient update. An understanding of these artefacts is important in the design of computationally affordable, good quality equalisers. A detailed investigation into these artefacts using various forms of arithmetic, filter frequency response, input excitation and sampling frequencies is described in this thesis. Novel coefficient calculation techniques, based on the matched z-transform (MZT) were developed to minimise filter response distortion and computation for on-line implementation. It was found that MZT-based filter responses can approximate more closely to s-plane filters, than BZTbased filters, with an affordable increase in computation load. Frequency response distortions and prewarping/correction schemes at higher sampling frequencies (96 and 192 kHz) were also assessed. An environment for emulating fractional quantisation in fixed and floating point arithmetic was developed. Various key filter topologies were emulated in fixed and floating point arithmetic using various input stimuli and frequency responses. The work provides detailed objective information and an understanding of the behaviour of key topologies in fixed and floating point arithmetic and the effects of input excitation and sampling frequency. Signal disturbance behaviour in key filter topologies during coefficient update was investigated through the implementation of various coefficient update scenarios. Input stimuli and specific frequency response changes that produce worst-case disturbances were identified, providing an analytical understanding of disturbance behaviour in various topologies. Existing parameter and coefficient interpolation algorithms were implemented and assessed under fihite wordlength arithmetic. The disturbance behaviour of various topologies at higher sampling frequencies was examined. The work contributes to the understanding of artefacts in audio equaliser implementation. The study of artefacts at the sampling frequencies of 48,96 and 192 kHz has implications in the assessment of equaliser performance at higher sampling frequencies.

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This work is concerned with encoding shape descriptors for a succession of the waveform segments to enable the transmission of speech signals at a low data rate. The segmentation was dependent on the identification of waveform features in speech signals thereby producing an irregular data rate from the time encoding process. The shape descriptors have been related to the real and complex zeros of a waveform through the theory of zero-based signal representation. A study of factors governing the data rates, the speech intelligibility and the buffer delay has been made for the above coding process based on waveform segmentation at zero-crossings. The redundancy in the average information conveyed by the zero-crossing data was investigated from conditional probability measurements resulting in the conclusion that a significant reduction in the data was available from coding procedures utilising the correlation in the data sequence. Signal pre-emphasis and dynamic range were found to control the segmentation rate, the variations in segmentation rate during an utterance determining the buffer size and delay. The transmission rate and the system delay necessary for time encoding were strongly influenced by the distortion arising from buffer management in matching the variable information rate to a constant transmission rate. A reduction by approximately a third in the transmission rate was observed to introduce data underflow distortion at a 200ms system delay setting into approximately 5% of the speech. Finally, a performance assessment of the time encoding process was made, subjectively by a reduced form of Diagnostic Rhyme Test (DRT) and objectively by spectral density plots comparisons. The results have indicated a data rate less than that for delta modulation and a processing complexity less than that for vocoders.

Helium speech is a term used to denote the speech produced by a deep sea diver breathing a Helium Oxygen mixture. The replacement of nitrogen in normal air by Helium solves some of the physiological problems associated with diving under pressure. However, it introduces severe distortion in diver's speech. The principal distortion is the nonlinear frequency expansion in the formant frequencies. A real time enhancement system has been constructed and partially tested. The design specification for this unscrambler has been generalised to enable the system to correct most of the Helium speech distortions. The system operates in the frequency domain and is based on the wide band analysis-synthesis technique. The system's algorithm for correcting the Helium speech distortion, is flexible and could be easily changed to satisfy different diving conditions. The possible use of the system to study Helium speech characteristics has also been considered.

This thesis is concerned primarily with the architecture of Digital Signal Computers. The work is supported by the design, development and application of a novel Digital Signal Computer system, the MAC68. The MAC68 is a Functional Multiprocessor, using two independent processors, one of which executes general-purpose tasks, and the other executes sequences of arithmetic. The particular MAC68 design was arrived at after careful evaluation of existing Digital Signal Computer architectures. MAC68 features are fully evaluated via its application to the Sub-Band Coding of speech, and in particular by the development of a 16Kb/s Sub-band Coder using six sub-bands. MAC68 performance was found to be comparable to that of current DSP micros for basic digital filter tasks, and superior for FFT tasks. The MAC68 architecture is a balance of high-speed arithmetic and general- purpose capabilities, and is likely to have a greater range of application than General-Purpose micros or DSP micros used alone. Suggestions are put forward for MAC68 enhancements utilising state-of-the-art hardware and software technologies. Because of the current widespread use of General-Purpose micros, and because of the possible performance gains to be had with the MAC68-type architecture, it is thought that MAC68 architectural concepts will be of value in the design of future high-performance Digital Signal Computer systems.

In time encoded speech (TES), information is transmitted relating to the distances between zero crossings and the shape of the waveform between successive zero crossings. The quality of the reconstructed TES signal will therefore depend on the accuracy to which these original signal parameters are presented in the reconstructed signal. When transmitting the waveform parameter descriptors (symbols), the variable TES symbol generation rate has to be matched to constant rate transmission channels using first-in first-out storage buffers. Since there are large variations in generation rates, at modest transmission rates, these buffers overflow destroying some of the symbols. Therefore in practical TES systems, the description of the original signal parameters will also depend on the amount of buffer distortion introduced in the transmission process. In this thesis, two techniques of describing the waveshape more: accurately than existing TES methods, four methods of controlling buffer overflow, and the auditory effects of these waveshape describing the buffer overflow control techniques are presented. Using the two new waveshape describing techniques and a parabolic reconstruction techniques it is shown that to obtain a significant improvement in quality in high quality TES Systems, a substantial increase in precise original signal information is required. Ways of achieving this kind of increase in original signal information without significantly increasing the data rate, has been suggested and demonstrated. Using the four buffer control strategies it is shown that for the control strategies to operate satisfactorily, buffer overflow in the voiced regions should be avoided. It is then shown that this can be achieved without significantly increasing the transmission rate, by exploiting properties of speech perception. Further, various methods of quantising TES parameters and the tradeoffs between quantisation and buffer overflow distortion are also investigated.

The work reported in this thesis deals with aspects of synchronisation and tracking in direct sequence spread spectrum systems used in ranging and communications applications. This is regarded as a major design problem in such systems and several novel solutions are presented. Three main problem areas have been defined: i) reduction of the acquisition time of code sychronisation in the spread spectrum receiver; ii) reduction of the receiver complexity; iii) improvement of the signal to noise ratio performance of the system by better utilisation of the power spectrum in the main lobe of the transmitted signal. Greater tolerance to Doppler shift effects is also important. A general review of the spread spectrum concept and past work is first given in Chapter One, and common methods of synchronisation and tracking are reviewed in Chapter Two. There, current performance limitations are also included. In Chapter Three a novel method is given for increasing the speed of synchronisation between locally generated and received codes, using a technique of controlling the loop's error curve during acquisition. This method is applied to different width delay lock loops, and a significant increase in maximum search rate is obtained. The effect of the width of the discriminator characteristics and damping ratio on the maximum search rate are also examined. The technique is applied to data modulated spread spectrum systems which use either synchronous or asynchronous data communication systems. All methods have been tested experimentally and found to perform as predicted theoretically. Several novel spread sprectrum configurations are given in Chapter Four which employ multi-level sequences. Some configurations have reduced the complexity and cost of the spread spectrum receivers. Others show some improvement in the maximum search rate as well as the signal to noise ratio performance. Some of these configurations have been implemented experimentally. In Chapter Five, the generation and properties of the composite (Kronecker) sequences are explained. Several types of component sequences are examined. And the reception of these composite sequences are discussed. In particular, a technique is introduced for achieving a rapid acquisition of phase synchronisation using these codes. The effect of white Gaussian noise on the acquisition performance of the delay lock loop is given in Chapter Six. Experimental results are obtained for both digital and analogue correlators. Chapter Seven gives a final summary of the conclusions, and further work suggestions.

Time Encoded Speech (TES) transmits information concerning the duration between zero-crossings, shape and the amplitude of the signal between successive zero-crossings. This thesis examines a number of aspects of TES with the view of achieving data reductions to enable the transmission of speech, with acceptable quality and intelligibility, at low bit rates and a practical system delay. This thesis presents: (i) A study of techniques for signalling amplitude information in a TES coder. It was indicated that a minimum of the order of 1 bit per epoch is required. Diagnostic Rhyme Tests (DRT) yielded intelligibility scores of the order of 88% for algorithms employing 1 and 2 bits of amplitude information per epoch. (ii) Investigations into Median and Moving Average filtering for preprocessing the epoch duration sequences. It has been shown that such applications, which involve simple numerical smoothing, are of little value for they degrade the quality of the synthesised speech. (iii) Studies of Extremal Coding and Orthogonal Transformations for achieving data reductions in the signalling of epoch duration and, in some instances, the peak magnitude sequences. Each technique yielded a useful data reduction. The technique using Hadamard Transformations yielded the greatest data reduction, a ratio of 2:1 for the representation of the epoch duration sequences. The Hadamard Transformation also proved to be of low complexity in its implementation. (iv) A real-time simplex digital voice channel, developed during the course of this thesis, and a study of the implementation of TES and TES related coders. It is reported that speech of acceptable quality and intelligibility is achieved for a transmission rate of 10 or 15kb/s with a transmission delay of 300ms.

Our primary mode of communication is via speech. Therefore, any person who has difficulty in producing understandable speech, for whatever reason, is at a great disadvantage. It is the role of the speech therapist to help such people to improve their speech production wherever possible. The aim of this work was to develop a computer-based speech therapy tutor for vowel production. The Tutor would be able to analyse monosyllabic utterances in real-time, extract the vowel portion and match this to a pre-determined template, and display the result with no appreciable delay. A fully-working prototype has been developed which employs general principles of aircraft tracking in a novel way, to track the coefficients of the quadratic factors of the all-pole linear-prediction model for speech production. It is shown how tracking these parameters can be used to extract extended vowels from a monosyllabic utterance. It is also shown how the algorithm which is used to determine the optimum frame-to-frame tracks can be used to perform template matching. The real plane on which the parameters are tracked, the rs-plane, suffers from non-linear scaling of frequency measures. This leads to poor spectral resolution of the perceptually-important low frequency parameters. To overcome this problem, the rs-plane can be warped in order that distance measures taken between points on the plane are more meaningful perceptually. The Tutor is based on a personal computer (PC). In order that real-time operation can be achieved, the processing power of the PC is enhanced by the addition of a digital signal processor (TMS32020) board and a transputer (T800) board. The prototype Tutor was developed with help and advice from Dundee Speech Therapy Service, Tayside Health Board, who also conducted a short pilot study of the Tutor.

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