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Wireless optical MIMO links can achieve very high data transmission rates by exploiting spatial diversity at a grand scale. Although such links can achieve high rates, their practical implementation remains challenging. The difficulty in implementation arises due to the complex transmitter and receiver designs required to overcome the channel impairments. This thesis considers practical transmitter and receiver designs for the wireless optical MIMO channel in the presence of channel impairments. In the first part of the thesis, two techniques to improve channel capacity for wireless optical MIMO channels are presented. The first technique uses multilevel half toning to reduce quantization noise power. For quantization noise-limited systems, increasing the number of quantizer levels provides gains in capacity. For example, at a rate of 200fps, a four-level quantizer gives approximately a two-fold increase in capacity over a binary-level quantizer for all frame sizes considered. The second technique uses higher order noise shaping to shape the quantization noise to the out-of-band spatial frequency spectrum. This technique is shown to be useful when the number of levels is small, i.e., near 2. In the second part of the thesis, the receiver design for wireless optical MIMO channels with magnification is considered. The work done in this part constitute a step towards the practical implementation of such links since it is the first time the effects of spatial transformations are considered. Signal magnification introduces varying spatial frequency inter-channel interference (SF-ICI) at the receiver. A novel receiver design that uses complex windowing with decision feedback equalization is used to equalize the SF-ICI in spatial frequency domain. For SF-ICI limited channels, the novel receiver design achieved a low bit-error rate compared with rectangular windowing with zero-forcing equalization. However, for noise limited channels, rectangular windowing with zero-forcing equalization is the receiver design of choice. / Thesis / Master of Applied Science (MASc)

Multimedia' nature of an application refers to the presence of several media streams in parallel. Whether it is receiving real-time data or retrieving stored data, there exists an end-to-end delay in data transfer from source to destination over the network. This delay experienced can be split into a fixed part and a variable part. Data processing time like coding and decoding at the source and destination are the fixed delays experienced. The variable delay occurs mainly due to queuing at the intermediate nodes during its flow through the network. The variable or unequal delays introduce gaps or discontinuities within a stream. In multi-stream applications where each stream may flow on different routes based on the bandwidth availability experiencing different delays, mismatch between them can also occur. These discontinuities and skews result in poor quality of playout. Clock drift and variations in drift rates between the source/s and destination/s, clock also lead to poor quality of play out. To eliminate these skews and discontinuities, there must be mechanisms, viz., and synchronisation services to convey, reintroduce and maintain the temporal relationship between the media streams for presentation throughout the playout, at the destination. The reintroduction of this lost temporal relationship within a stream and between various media streams for presentation at the destination is the object of multimedia synchronisation and is the subject matter of this thesis. In the presence of synchronised clocks, the main cause of asynchronies between media streams is the difference in delays experienced and the jitter. In this work, to convey the temporal relationship between streams of an application to the playout site, each stream is assigned a priority л, based on its importance to the user. The media streams are then divided into synchronisation units called 'Groups' based on that stream's characteristics which has the highest priority л. A group may therefore consist of one video frame and other data which were generated in that interval. Or may consist of silence and talk-spurt periods of the voice stream with data units of other streams generated in the same interval. Since the quality of playout of temporally related delay-sensitive streams depends upon the delay-experienced, the concept of QoS can be extended to describe the presentation requirements of uch data. Depending on the user perception and the delay experienced, an application can have a range of playout times, giving the best performance. The presentation of many real-time applications can be considered satisfactory even when the delay bound is exceeded by a small amount for a short period of time under varying network conditions. This property can be exploited by defining two sets of QoS parameters, namely QoS optimum and QoSlimit for each real-time application. As the delay and its variations increase, the optimum playout time range decreases. QoS optimum specifies the performance parameters required to perceive 'realtime'. Multimedia data can be played out at its QoSlimit with a deterioration in quality under poor network conditions still maintaining the synchronisation between streams. To control the playout at two levels of QoS, and maintain intra-media and inter-media synchronisation, stream controllers and super stream controllers have been used. The dynamic delay compensation algorithm and synchronisation services were simulated using network delay models and performances studied. It is shown that the proposed algorithm not only synchronised media streams and smoothened jitter but also optimised buffer space and buffer occupancy time while meeting the desired quality of service requirements

Thesis (M.B.A.)--University of Hong Kong, 1997. / Includes bibliographical references (leaf 56).

published_or_final_version / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

published_or_final_version / abstract / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

The object of interest in this research is the evaluation of the combined routing and flow control scheme proposed by Muralidhar and Sundareshan. A slightly modified design using results from positive system theory is proposed as an attempt to improve the performance. Then a detailed performance evaluation of Muralidhar and Sundareshan's scheme and its modified version proposed here is conducted using discrete-event simulation. The application of positive system theory to the combined routing and flow control algorithm proposed by Muralidhar and Sundareshan proved to be quite effective in improving the overall performance only in the initial transient period. The fact that the improvement is restricted to a particular time period appears to be attributable to poor buffer management and certain other fundamental problems with Muralidhar and Sundareshan's scheme. These problems are identified and are described in detail.

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