Academic literature on the topic 'Fields of Research – 290000 Engineering and Technology – 291700 Communications Technologies'

Currently, digital signal processing systems typically assume that the signals are bandlimited. This is due to our knowledge based on the uniform sampling theorem for bandlimited signals which was established over 50 years ago by the works of Whittaker, Kotel'nikov and Shannon. However, in practice the digital signals are mostly of finite length. This kind of signals are not strictly bandlimited. Furthermore, advances in electronics have led to the use of very wide bandwidth signals and systems, such as Ultra-Wide Band (UWB) communication systems with signal bandwidths of several giga-hertz. This kind of signals can effectively be viewed as having infinite bandwidth. Thus there is a need to extend existing theory and techniques for signals of finite bandwidths to that for non-bandlimited signals. Two recent approaches to a more general sampling theory for non-bandlimited signals have been published. One is for signals with finite rate of innovation. The other introduced the concept of consistent sampling. It views sampling and reconstruction as projections of signals onto subspaces spanned by the sampling (acquisition) and reconstruction (synthesis) functions. Consistent sampling is achieved if the same discrete signal is obtained when the reconstructed continuous signal is sampled. However, it has been shown that when this generalized theory is applied to the de-interlacing of video signals, incorrect results are obtained. This is because de-interlacing is essentially a resampling problem rather than a sampling problem because both the input and output are discrete. While the theory for the resampling for bandlimited signals is well established, the problem of resampling without bandlimited constraints is largely unexplored. The aim of this thesis is to develop a resampling theory for non-bandlimited discrete signals and explore some of its potential applications. The first major contribution is the the theory and techniques for designing an optimal resampling system for signals in the general Hilbert Space when noise is not present. The system is optimal in the sense that the input of the system can always be obtained from the output. The theory is based on the concept of consistent resampling which means that the same continuous signal will be obtained when either the original or the resampled discrete signal is presented to the reconstruction filter. While comparing the input and output of a sampling/reconstruction system is relatively simple since both are continuous signals, comparing the discrete input and output of a resampling system is not. The second major contribution of this thesis is the proposal of a metric that allows us to evaluate the performance of a resampling system. The performance is analyzed in the Fourier domain as well. This performance metric also provides a way by which different resampling algorithms can be compared effectively. It therefore facilitates the process of choosing proper resampling schemes for a particular purpose. Unfortunately consistent resampling cannot always be achieved if noise is present in the signal or the system. Based on the performance metric proposed, the third major contribution of this thesis is the development of procedures for designing resampling systems in the presence of noise which is optimal in the mean squared error (MSE) sense. Both discrete and continuous noise are considered. The problem is formulated as a semi-definite program which can be solved effciently by existing techniques. The usefulness and correctness of the consistent resampling theory is demonstrated by its application to the video de-interlacing problem, image processing, the demodulation of ultra-wideband communication signals and mobile channel detection. The results show that the proposed resampling system has many advantages over existing approaches, including lower computational and time complexities, more accurate prediction of system performances, as well as robustness against noise.

This thesis provides a thorough analysis of the theoretical foundations and properties of the Spectral Warping Transform. The spectral warping transform is defined as a time-domain-to-time-domain digital signal processing transform that shifts the frequency components of a signal along the frequency axis. The z -transform coefficients of a warped signal correspond to z -domain ‘samples’ of the original signal that are unevenly spaced along the unit circle (equivalently, frequency-domain coefficients of the warped signal correspond to frequency-domain samples of the original signal that are unevenly spaced along the frequency axis). The location of these unevenly spaced frequency-domain samples is determined by a z -domain mapping function. This function may be arbitrary, except that it must map the unit circle to the unit circle. It is shown that, in addition to the frequency location, the bandwidth, duration and amplitude of each frequency component of a signal are affected by spectral warping. Specifically, frequency components within bands that are expanded in frequency have shortened durations and larger amplitudes (conversely, components in compressed frequency bands become longer with smaller amplitudes). A property related to the expansion and compression of the duration of frequency components is that if a signal is time delayed (its digital sequence is prepended with zeroes) then each of the frequency components will have a different delay after warping. This time-domain separation phenomenon is useful for separating in time the frequency components of a signal. Such separation is employed in the generation of spectrally flat chirp signals. Because spectral warping will generally expand the duration of some frequency components within a signal, the transform must produce more output samples than there are (non-zero) input samples in order to avoid time-domain aliasing. A discussion of the necessary output signal length is presented. Particular attention is given to spectral warping using all-pass mapping function, which can be realised as a cascade of all-pass filters. There exists an efficient hardware implementation for this all-pass SW realisation [1, 2]. A proof-of-concept application-specific integrated circuit that performs the core operations required by this algorithm was developed. Another focus of the presented research is spectral warping using a piecewise- linear mapping function. This type of spectral warping has the advantage that the changes in frequency, duration and amplitude between the non-warped and warped signals are constant factors over fixed frequency bands. A matrix formulation of the spectral warping transformation is developed. It presents the spectral warping transform as a single matrix multiplication. The transform matrix is the product of the three matrices that represent three conceptual steps. The first step is to apply a discrete Fourier transform to the time-domain signal, providing the frequency-domain representation. Step two is an interpolation to produce the signal content at the desired new frequency samples. This interpolation effectively provides the frequency warping. The final step is an inverse DFT to transform the signal back into the time domain. A special case of the spectral warping transform matrix has the same result as a linear (finite-impulse-response) filter, showing that spectral warping is a generalisation of linear filtering. The conditions for the invertibility of the spectral warping transformation are derived. Several possible realisation of the SW transform are discussed. These include two realisation using parallel finite-impulse-response filter banks and a realisation that uses a cascade of infinite-impulse-response filters. Finally, examples of applications for the spectral warping transform are given. These include: non-uniform spectral analysis (and signal generation), approximate spectral analysis in the time domain, and filter design. This thesis concludes that the SW transform is a useful tool for the manipulation of the frequency content of digital signals, and is particularly useful when the frequency content of a signal (or the frequency response of a system) over a limited band is of interest. It is also claimed that the SW transform may have valuable applications for embedded mixed-signal testing.

The growth of interactive multimedia (IMM) applications is one of the major driving forces behind the swift evolution of next-generation wireless networks where the traffic is expected to be varying and widely diversified. The amalgamation of multimedia applications on high-speed wireless networks is somewhat a natural evolution. Wireless local area network (WLAN) was initially developed to carry non-real time data. Since this type of traffic is bursty in nature, the channel access schemes were based on contention. However real time traffic (e.g. voice, video and other IMM applications) are different from this traditional data traffic as they have stringent constraints on quality of service (QoS) metrics like delay, jitter and throughput. Employing contention free channel access schemes that are implemented on the point coordination function (PCF), as opposed to the numerous works on the contending access schemes, is the plausible and intuitive approach to accommodate these innate requirements. Published researches show that works have been done on improving the distributed coordination function (DCF) to handle IMM traffic. Since the WLAN traffic today is a mix of both, it is only natural to utilize both, DCF and PCF, in a balanced manner to leverage the inherent strengths of each of them. We saw a scope in this technique and develop a scheme that combines both contention and non-contention based phases to handle heterogeneous traffic in WLAN. Standard access scheme, like 802.11e, improves DCF functionality by trying to emulate the functions of PCF. Researchers have made a multitude of improvements on 802.11e to reduce the costs of implementing the scheme on WLAN. We explore improving the PCF, instead, as this is more stable and implementations would be less costly. The initial part of this research investigates the effectiveness of the point coordination function (PCF) for carrying interactive multimedia traffic in WLAN. The performance statistics of IMM traffic were gathered and analyzed. Our results showed that PCF-based setup for IMM traffic is most suitable for high load scenarios. We confirmed that there is a scope in improving IMM transmissions on WLAN by using the PCF. This is supported by published researches on PCF related schemes in carrying IMM traffic on WLAN. Further investigations, via simulations, revealed that partitioning the superframe (SF) duration according to the need of the IMM traffic has considerable impact on the QoS of the WLAN. A theoretical model has been developed to model the two phases, i.e., PCF and DCF, of WLAN medium access control (MAC). With this model an optimum value of the contention free period (CFP) was calculated to meet the QoS requirement of IMM traffic being transmitted. Treating IMM traffic as data traffic or equating both IMM and non-IMM together could compromise a fair treatment that should be given to these QoS sensitive traffic. A self-adaptive scheme, called MAC with Dynamic Superframe Selection (MDSS) scheme, generates an optimum SF configuration according to the QoS requirements of traversing IMM traffic. That particular scheme is shown to provide a more efficient transmission on WLAN. MDSS maximizes the utilization of CFP while providing fairness to contention period (CP). The performance of MDSS is compared to that of 802.11e, which is taken as the benchmark for comparison. Jitter and delay result for MDSS is relatively lower while throughput is higher. This confirms that MDSS is capable of making significant improvement to the standard access scheme.

The quality of Machine Translation (MT) can often be poor due to it appearing incoherent and lacking in fluency. These problems consist of word ordering, awkward use of words and grammar, and translating text too literally. However we should not consider translations such as these failures until we have done our best to enhance their quality, or more simply, their fluency. In the same way various processes can be applied to touch up a photograph, various processes can also be applied to touch up a translation. This research outlines the improvement of MT quality through the application of Fluency Enhancement (FE), which is a process we have created that reforms and evaluates text to enhance its fluency. We have tested our FE process on our own MT system which operates on what we call the SAM fundamentals, which are as follows: Simplicity - to be simple in design in order to be portable across different languages pairs, Adaptability - to compensate for the evolution of language, and Multiplicity - to determine a final set of translations from as many candidate translations as possible. Based on our research, the SAM fundamentals are the key to developing a successful MT system, and are what have piloted the success of our FE process.