Allocating resources to variable bitrate (VBR) teletraffic sources is not a trivial task because the impact of such sources on a buffered switch is difficult to predict. This problem has repercussions for call admission control (CAC) on asynchronous transfer mode (ATM) networks. In this thesis we report on investigations into the nature of several types of VBR teletraffic. The purpose of these investigations is to identify parameters of the traffic that may assist in the development of CAC algorithms. As such we concentrate on the correlation structure and marginal distribution; the two aspects of a teletraffic source that affect its behaviour through a buffered switch. The investigations into the correlation structure consider whether VBR video is selfsimilar or non-stationary. This question is significant as the exponent of self-similarity has been identified as being useful for characterising VBR teletraffic. Although results are inconclusive with regards to the original question, they do show that self-similar models are best able to capture the video data's behaviour. The investigations into the marginal distributions are in two parts. The first considers applying a structured Markovian model to ATM data and demonstrates how model parameters can be estimated from measurable properties of teletraffic data. This has implications for parametric CAC. The second part considers the use of stable distributions in teletraffic characterisation and modelling. We show that several teletraffic datasets are heavy tailed and then develop a framework for the estimation of stable distribution parameters. We finish by considering the effective bandwidths of stable distributions and models and by considering the effect of stable parameters on model behaviour. This is done in an attempt to develop a CAC algorithm based on the paradigms of self-similarity and stable distributions.

Digital visual communication has been increasingly adopted as an efficient new medium in a variety of different fields; multi-media computers, digital televisions, telecommunications, etc. Exchange of visual information between remote sites requires that digital video is encoded by compressing the amount of data and transmitting it through specified network connections. The compression and transmission of digital video is an amalgamation of statistical data coding processes, which aims at efficient exchange of visual information without technical barriers due to different standards, services, media, etc. It is associated with a series of different disciplines of digital signal processing, each of which can be applied independently. It includes a few different technical principles; distortionrate theory, prediction techniques and control theory. The MPEG (Moving Picture Experts Group) video compression standard is based on this paradigm, thus, it contains a variety of different coding parameters which may result in different performance depending on their values. It specifies the bit stream syntax and the decoding process as its normative parts. The encoder details remain nonnormative and are configured by a specific design. This means that the MPEG video encoder has a great deal of flexibility in the aspects of performance and implementation. This thesis deals with control techniques for the data rate of compressed video, which determine the encoding efficiency and video quality. The video rate control is achieved by adjusting quantisation step size depending on the occupancy of a transmission buffer memory which stores the compressed video data for a specific period of time. Conventional video rate control techniques have generally been based either on linear predictive or on control theoretic models. However, this thesis takes a different view on digital video and MPEG video coding, and focuses on the non-stationary and nonlinear nature of realistic moving pictures. Furthermore, considering the MPEG encoding structure involved in the different disciplines, A series of improvements for video rate control are proposed, each of which enhances the performance of the MPEG encoder. A nonlinear quantisation control technique is investigated, which controls the buffer occupancy with the quantiser using a series of nonlinear functions. Linear and nonlinear feed-forward networks are also employed to control the quantiser. The linear combiner is used as a linear estimator and a radial basis function network as a nonlinear one. Finally, fuzzy rulebased control is applied to exploit the advantages of the nonlinear control technique which is able to provide linguistic judgement in the control mechanism. All these techniques are employed according to two global approaches (feedforward and feedback) applied to the rate control. The performance evaluation is carried out in terms of controllability over bit rate variation and video quality, by conducting a series of simulations.

With the emergence of nonlinear dynamical systems analysis over recent years it has become clear that conventional time domain and frequency domain approaches to speech synthesis may be far from optimal. Using state space reconstructions of the time domain speech signal it is, at least in theory, possible to investigate a number of invariant geometrical measures for the underlying system which give a more thorough understanding of the dynamics of the system and therefore the form that any model should take. This thesis introduces a number of nonlinear dynamical analysis tools which are then applied to a database of vowels to extract the underlying invariant geometrical properties. The results of this analysis are then applied, using ideas taken from nonlinear dynamics, to the problem of speech synthesis and a novel synthesis technique is described and demonstrated. The tools used for the analysis are time delay embedding, singular value decomposition, correlation dimension, local singular value analysis, Lyapunov spectra and short term prediction properties. Although there have been many papers written about these tools, and algorithms proposed, there are currently no generally accepted techniques, especially for the calculation of Lyapunov spectra in the presence of noise and data length limitations. This thesis introduces all of the above tools and looks in detail at Lyapunov exponents and two major novel modifications are proposed that are demonstrated to be more robust than conventional techniques. The novel robust techniques are applied to a large database of vowel sounds showing that the vowels tested show evidence of nonlinear, low-dimensional, non-chaotic behaviour. It is particularly the evidence of non-chaotic behaviour that is of importance from a synthesis point of view and is used in the final section of the thesis which introduces a novel synthesis technique. The synthesis technique, which is based on ideas taken from nonlinear dynamics theory is detailed and demonstrated showing that it is capable of high quality natural sounding speech.

This thesis describes the application of a perceptually based spectral subtraction algorithm for the enhancement of non-stationary noise corrupted speech. Through examination of speech enhancement techniques, explanations are given for the choice of magnitude spectral subtraction and how the human auditory system can be modelled for frequency domain speech enhancement. It is discovered, that the cochlea provides the mechanical speech enhancement in the auditory system, through the use of masking. Frequency masking is used in spectral subtraction, to improve the algorithm execution time, and to shape the enhancement process making it sound natural to the ear. A new technique for estimation of background noise is presented, which operates during speech sections as well as pauses. This uses two microphones placed on opposite ends of the cellular handset. Using these, the algorithm determines whether the signal is speech, or noise, by examining the current and next frames presented to each microphone. This allows operation in non-stationary conditions, as the estimation is calculated for each frame, and a speech pause is not required for updating. A voting decision process decides the presence of speech or noise which determines which microphone the estimation is calculated from. The importance of an accurate noise estimate is highlighted with a new technique to reduce the effect of musical noise artifacts in the processed speech. This is a classic drawback of spectral subtraction techniques, and it is shown, that the trade off between noise reduction and speech distortion can be extended by this process. A new method for dealing with musical noise is described, which uses a combination of energy and variance examination of the spectrogram to segregate potential musical noise from desired speech sections. By examination of the spectrogram points surrounding musical noise sections, perceptually relevant values replace the corruption leading to cleaner enhanced speech. Any perceptual speech system requires accurate estimates of the clean speech masking thresholds, to prevent noisy sections being passed through the enhancement untouched. In this thesis, a method for the calculation of the estimated clean speech masking thresholds is derived. Classically, this requires an estimation of the clean speech before the thresholds can be derived, but this results in inaccuracy due to the presence of musical noise and spectral nulls. The new algorithm examines the thresholds produced by the corrupted speech, and the background noise, and from these determines the relationship between the two, to produce an estimate of the clean thresholds, with no operation performed on the actual speech signal. A discrepancy is found between the results for male and female speech, which, by examination of the perceptual process, is shown to be due to the different formant positions in male and female speech. Following the development of these parts, the entire enhancement algorithm is tested on a range of noise scenarios, using male and female speech. The results show, that the proposed algorithm is able to provide adequate performance in terms of noise reduction and speech quality.

Speech synthesis technology plays an important role in many aspects of man–machine interaction, particularly in telephony applications. In order to be widely accepted, the synthesised speech quality should be as human–like as possible. This thesis investigates novel techniques for the speech signal generation stage in a speech synthesiser, based on concepts from nonlinear dynamical theory. It focuses on natural–sounding synthesis for voiced speech, coupled with the ability to generate the sound at the required pitch. The one–dimensional voiced speech time–domain signals are embedded into an appropriate higher dimensional space, using Takens’ method of delays. These reconstructed state space representations have approximately the same dynamical properties as the original speech generating system and are thus effective models. A new technique for marking epoch points in voiced speech that operates in the state space domain is proposed. Using the fact that one revolution of the state space representation is equal to one pitch period, pitch synchronous points can be found using a Poincar´e map. Evidently the epoch pulses are pitch synchronous and therefore can be marked. The same state space representation is also used in a locally–linear speech synthesiser. This models the nonlinear dynamics of the speech signal by a series of local approximations, using the original signal as a template. The synthesised speech is natural–sounding because, rather than simply copying the original data, the technique makes use of the local dynamics to create a new, unique signal trajectory. Pitch modification within this synthesis structure is also investigated, with an attempt made to exploit the ˇ Silnikov–type orbit of voiced speech state space reconstructions. However, this technique is found to be incompatible with the locally–linear modelling technique, leaving the pitch modification issue unresolved. A different modelling strategy, using a radial basis function neural network to model the state space dynamics, is then considered. This produces a parametric model of the speech sound. Synthesised speech is obtained by connecting a delayed version of the network output back to the input via a global feedback loop. The network then synthesises speech in a free–running manner. Stability of the output is ensured by using regularisation theory when learning the weights. Complexity is also kept to a minimum because the network centres are fixed on a data–independent hyper–lattice, so only the linear–in–the–parameters weights need to be learnt for each vowel realisation. Pitch modification is again investigated, based around the idea of interpolating the weight vector between different realisations of the same vowel, but at differing pitch values. However modelling the inter–pitch weight vector variations is very difficult, indicating that further study of pitch modification techniques is required before a complete nonlinear synthesiser can be implemented.

The growing demand for capacity in wireless communications is the driving force behind improving established networks and the deployment of a new worldwide mobile standard. Capacity calculations show that the direct sequence code division multiple access (DS-CDMA) technique has more capacity than the time division multiple access technique. Therefore, most 3rd generation mobile systems will incorporate some sort of DS-CDMA. In this thesis DS-CDMA receiver structures are investigated from the view point of pattern recognition which leads to new DS-CDMA receiver structures. It is known that the optimum DS-CDMA receiver has a nonlinear structure with prohibitive complexity for practical implementation. It is also known that the currently implemented receiver in 2nd generation DSCDMA mobile handsets has poor performance, because it suffers from multiuser interference. Consequently, this work focuses on sub-optimum nonlinear receivers for DS-CDMA in the downlink scenario. First, the thesis reviews DS-CDMA, established equalisers, DS-CDMA receivers and pattern recognition techniques. Then the new receivers are proposed. It is shown that DS-CDMA can be considered as a pattern recognition problem and hence, pattern recognition techniques can be exploited in order to develop DS-CDMA receivers. Another approach is to apply known equaliser structures for DS-CDMA. One proposed receiver is based on the Volterra series expansion and processes the received signal at the chip rate. Another receiver is a symbol rate radial basis function network (RBFN) receiver with reduced complexity. Subsequently, a receiver is proposed based on linear programming (LP) which is especially tailored for nonlinearly separable scenarios. The LP based receiver performance is equivalent to the known decorrelating detector in linearly separable scenarios. Finally, a hybrid receiver is proposed which combines LP and RBFN and which exploits knowledge gained from pattern recognition. This structure has lower complexity than the full RBF and good performance, and has a large potential for further improvements. Monte-Carlo simulations compare the proposed DS-CDMA receivers against established linear and nonlinear receivers. It is shown that all proposed receivers outperform the known linear receivers. The Volterra receiver’s complexity is relatively high for the performance gain achieved and might not suit practical implementation. The other receiver’s complexity was greatly reduced but it performs nearly as well as an optimum symbol by symbol detector. This thesis shows that DS-CDMA is a pattern recognition problem and that pattern recognition techniques can simplify DS-CDMA receiver structures. Knowledge is gained from the DSCDMA signal patterns which help to understand the problem of a DS-CDMA receiver. It should be noted that from the large number of known techniques, only a few pattern recognition techniques are considered in this work, and any further work should look at other techniques. Pattern recognition techniques can reduce the complexity of existing DS-CDMA receivers while maintaining performance, leading to novel receiver structures.

The main subject of this thesis is the investigation of antenna array techniques for improving the performance of the downlink of wideband code division multiple access (WCDMA) mobile communication systems. These communication systems operate in frequency division duplex (FDD) mode and the antenna arrays are employed in the base station. A number of diversity, beamforming and hybrid techniques are analysed and their bit error ratio (BER) versus signalto- noise ratio (SNR) performance is calculated as a function of the eigenvalues of the mean channel correlation matrix, where this is applicable. Also, their BER versus SNR performance is evaluated by means of computer simulations in various channel environments and using different numbers of transmit antenna elements in the base station. The simulation results of the techniques, along with other characteristics, are compared to examine the relationship among their performance in various channel environments and investigate which technique is most suitable for each channel environment. Next, a combination of the channel correlation matrix eigenvalue decomposition and space-time processing is proposed as a possible open loop approach to the downlink data signal transmission. It decomposes the channel into M components in the form of eigenvectors (M is the number of transmit antennas in the base station), and attempts to minimise the transmit power that is needed to achieve a target BER at the mobile receiver by employing the optimum number of these eigenvectors. The lower transmit power and the directional transmission by means of eigenvectors are expected to lower interference levels to non-desired users (especially to those users who are not physically close to the direction(s) of transmission). Theoretical and simulation results suggest that this approach performs better than other presented open loop techniques, while the performance gain depends on M and the channel environment. In simulations it is usually assumed that the base and mobile station have access to perfect estimates of all needed parameters (e.g. channel coecients). However, in practical systems they make use of pilot and/or feedback signals to obtain estimates of these parameters, which result in noisy estimates. The impact of the noisy estimates on the performance of various techniques is investigated by computer simulations, and the results suggest that there is typically some performance loss. The loss depends on the parameter that is estimated from pilot signals, and may be a function of M, SNR and/or the channel environment. In certain beamforming techniques the base station operates the transmit antenna array in an open loop fashion by estimating the downlink weight vector from the directional information of the uplink channel. Nevertheless, in FDD systems this results in performance loss due to the separation between the uplink and downlink carrier frequencies (`FDD gap'). This loss is quantified and the results show that it is a function of M and the FDD gap. Also, a very simple technique for compensating this loss is proposed, and results obtained after its application suggest that it eliminates most of the loss. Comparison of the proposed technique with an existing compensation technique suggests that, even though the latter is more complex than the former, it yields very little additional improvement.

Techniques which utilise a signal's Higher Order Statistics (HOS) can reveal information about non-Gaussian signals and nonlinearities which cannot be obtained using conventional (second-order) techniques. This information may be useful in speech processing because it may provide clues about how to construct new models of speech production which are better than existing models. There has been a recent surge of interest in the application of HOS techniques to speech processing, but this has been handicapped by a lack of understanding of what the HOS properties of speech signals are. Without this understanding the HOS information which is in speech signals can not be efficiently utilised. This thesis describes an investigation into the use of HOS techniques, in particular the third-order frequency domain measure called the bispectrum, to speech signals. Several issues relating to bispectral speech analysis are addressed, including nonlinearity detection, pitch-synchronous analysis, estimation criteria and stationarity. A flaw is identified in an existing algorithm for detecting quadratic nonlinearities, and a new detector is proposed which has better statistical properties. In addition, a new algorithm is developed for estimating the normalised bispectrum of signals contaminated by transient noise. Finally the tools developed in the study are applied to a specially constructed database of continuant speech sounds. The results are consistent with the hypothesis that speech signals do not exhibit quadratic nonlinearity.

The introduction of cellular wireless systems in the 1980s has resulted in a continuous and growing demand for personal communication services. This demand has made larger capacity systems necessary. With the interest from both the research community and industry in wireless code-division multiple-access (CDMA) systems, the application of multiuser detection (MUD) techniques to wireless systems is becoming increasingly important. MUD is an important area of interest to help obtain the significant increase in capacity needed for future wireless services. The standardisation of direct-sequence CDMA (DS-CDMA) systems in the third generation of mobile communication systems has raised even more interest in exploiting the capabilities and capacity of this type of technology. However, the conventional DS-CDMA system has the major problem of multiple-access interference (MAI). The MAI is unavoidable because receivers deal with information which is transmitted not by a single source but by several uncoordinated and geographically separated sources. As a result, the capacity of these systems is inherently interference limited by other users. To overcome these limitations, MUD emerges as a promising approach to increase the system capacity. This thesis addresses the problem of improving the downlink capacity of a coded DS-CDMA system with the use of MUD techniques at the mobile terminal receiver. The optimum multiuser receiver scheme is far too computational intensive for practical use. Therefore, the aim of this thesis is to investigate sub-optimal multiuser receiver schemes that can exploit the advantages of MUD but also simplify its implementation. The attention is primarily focused on iterative MUD receiver schemes which apply the turbo multiuser detection principle. Essentially this principle consists of an iterative exchange of extrinsic information among the receiver modules to achieve improved performance. In this thesis, the implementation of an iterative receiver based on a linear MUD technique and a cancellation scheme over an additive white Gaussian noise (AWGN) channel is first proposed and analysed. The interference analysis shows that good performance is achieved using a lowcomplexity receiver structure. In more realistic mobile channels, however, this type of receiver suffers from the presence of higher levels of interference resulting in poor receiver performance. The reason for this is that in such scenarios the desired signals are no longer linearly separable. Therefore, non-linear detection techniques are required to provide better performance. With this purpose, a hybrid iterative multiuser receiver is investigated for the case of a stationary multipath channel. The incorporation of antenna arrays is an effective and practical technique to provide a significant capacity gain over conventional single-antenna systems. In this context, a novel space-time iterative multiuser receiver is proposed which achieves a large improvement in spectral efficiency and performance over multipath fading channels. In addition, it is shown that this architecture can be implemented without a prohibitive complexity cost. The exploitation of the iterative principle can be used to approach the capacity bounds of a coded DS-CDMA system. Using the Shannon’s sphere packing bound, a comparison is presented to illustrate how closely a practical system can approach the theoretical limits of system performance.

In this work we consider homogeneous continua X with the property that $\check H\sp1(X,\doubz)\ne 0$, construct a covering space $\tilde X$ of X and study some of its properties. We describe $\tilde X$ for several specific continua / acase@tulane.edu