Thesis (Ph.D.) -- University of Maryland, College Park, 2007. / Thesis research directed by Dept. of Philosophy. Originally produced as a documentary film in 2007. Includes trailer (4 min.). Booklet includes a preface, explanatory text, discussion questions, suggested reading, and a bibliography. Appendix 1 concerns the film; it includes a scene-by-scene synopsis, song lyrics (in Spanish & English), and a transcript of the film (in Spanish, with a side-by-side transcript of the English subtitles).

Thesis (M.A.)--York University, 2004. Graduate Programme in Communication. / Typescript. Includes bibliographical references (leaves 112-116). Also available on the Internet. MODE OF ACCESS via web browser by entering the following URL: http://gateway.proquest.com/openurl?url%5Fver=Z39.88-2004&res%5Fdat=xri:pqdiss &rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&rft_dat=xri:pqdiss:MR11869

Thesis (Ph. D.)--University of Washington, 2000. / Vita. Includes bibliographical references (p. 110-119).

The subject of this dissertation is coherent digital acoustic communication in the underwater environment. The objective of the research is to develop algorithms for reliable communication in the shallow underwater channel. Investigation is focused on channel depth less than 100 m and distances between transmitter and receiver from 5 km to 50 km. Based on the characteristics of the underwater acoustic channel and using a conventional approach, the achievable transmission range and the required acoustic power is determined for given channel conditions and system parameters. A channel model suitable for the investigation of shallow water communication is developed which takes into account transmitter-receiver geometry, environmental conditions and system parameters. The model is based on multiple reflections in the channel with weightings according to signal attenuation due to spreading, reflection losses and absorption. Time-variability is introduced by incorporating Doppler frequency shifts due to transmitter/receiver motion. A new method of evaluating performance of a system operating in such multipath conditions is proposed by introduction of a signal-to-multipath ratio (SMR), which is a measure of intersymbol interference (ISI) caused by the multipath. The SMR allows assessment of system performance for various receiver/transmitter positions and channel parameters. It can be used, for instance, to find the transmitter/receiver depth for optimum transmission. A suitable equalizer can improve a SMR. For example, a decision feedback equalizer (DFE) using a least mean square (LMS) and fast optimized LMS criterion is effective in coping with ISI as demonstrated by computer simulations. Hardware complexities of several equalizer algorithms are investigated for a selected channel. The performance degradation due to the presence of Gaussian noise in addition to multipath is analyzed by simulation. A novel structure of an equalizer suitable for the time-variant underwater acoustic channel is proposed. By adaptively adjusting the number of equalizer taps depending on the channel condition, the proposed structure offers reduced hardware complexity. Computer simulations demonstrate the effectiveness of this approach. It is anticipated that the results of this work will find application in the design of high data rate transmission systems for ocean bottom instrumentation, the design of telemetry for autonomous underwater vehicles, and others. / Graduate

The past two decades have witnessed a rapid expansion within the telecommunications industry. This growth has been primarily motivated by the proliferation of digital communication systems and services which have become easily available through wired and wireless systems. Current research trends involve the integration of speech, audio, video and data channels into true multimedia communications over fixed and mobile networks. However, while the available bandwidth in wired terrestrial networks is relatively cheap and expandable, it becomes a limited resource in satellite and cellular-radio systems. In order to accommodate an ever growing number of users while maintaining high quality and low operational costs, it is necessary to maximise spectral efficiency. This has given rise to the development of high rate compression techniques with the ability to adapt to a broad class of input signals and to varying network resources. The research carried out in this thesis has mainly focused on the design of a single algorithm for compressing speech and audio signals sampled at different rates. The algorithms are based on the analysis-by-synthesis linear prediction coding (AbS-LPC) scheme, which has been widely employed in various speech coding standards. However, this bit rate reduction technique is based on the speech production mechanism and as such provides a rigid structure which presents a major limitation for audio coding. In order to improve the audio quality at low rates and to compensate for the errors incurred by the linear prediction during segments of high transitions, the algorithms employ an efficient pulse excitation structure which represents the short innovation sequences with sparse unit magnitude pulses. The scheme proposed for the compression of telephone bandwidth speech and audio signals at 12kb/s achieves similar quality to the G.728 coder at 16kb/s and higher audio quality than the GSM-EFR standard at 12.2kb/s. Wideband speech and audio coding schemes have been designed using both the fullband approach at bit rates of 17 and 19kb/s and also the split band technique at a bit rate of 20kb/s. The perceptual quality is comparable to the G.722 coder operating at 48kb/s. The subband decomposition technique is also adapted to code speech and audio signals sampled at 32kHz. The quality of the coder at 28kb/s is similar to the quality achieved by the MP3 coder at 32kb/s. The algorithm also provides bandwidth and bit rate scalability ranging from 12 to 64kb/s, making it ideal for deployment in rate-adaptive communication systems.

D.Ing. / When digital information is to be transmitted over a communications channel or stored in a data recording system, it is first mapped onto a code sequence by an encoder. The code sequence has certain properties which makes it suitable for use on the channel, ie the sequence complies to the channel input restrictions. These input restrictions are often described in terms of a required power spectral density of the code sequence. In addition, the code sequence can also be chosen in such a way as to enable the receiver to correct errors which occur in the channel. The set of rules which governs the encoding process is referred to as a line code or a modulation code for the transmission or storage of data, respectively. Before a new line code or modulation code can be developed, the properties that the code sequence should have for compliance to the channel input, restrictions and possession of desired error correction capabilities have to be established. A code' construction algorithm, which is often time consuming and difficult to apply, is then used to obtain the new code. In this dissertation, new classes of sequences which comply to the input restrictions and error correction requirements of practical channels are defined, and new line codes and recording codes are developed for mapping data onto these sequences. Several theorems which show relations between' information theoretical aspects of different classes of code sequences are presented. Algorithms which can be used to transform an existing line code or modulation code into a new code for use on another channel are introduced. These algorithms are systematic and easy to apply, and precludes the necessity of applying a code construction algorithm.

No description available.

Location privacy preservation has shifted to the forefront of discussions about next generation wireless networks. While pseudonym-changing schemes have been proposed to preserve an individual's privacy, simulation has shown that new association attack models render these schemes useless. The major contribution of this thesis is the implementation of a tracking network with commodity hardware on the California Polytechnic State University campus which leverages the combination of de-anonymization strategies on captured wireless network data to show the effectiveness of a pseudonym-changing scheme for wireless identification across WiFi and Zigbee protocols.

Quadrature phase shift keying (QPSK) and minimum shift keying (MSK) are the two most used M-ary modulation techniques in Direct-Sequence (DS) Spread Spectrum Communication systems. This thesis introduces a new modulation technique that can compete well with QPSK and MSK in many applications. This new modulation technique, made up of a superposition of one QPSK signal and two amplitude weighted QPSK signals, is called Amplitude Weighted Quadrature Phase Shift Keying (AWQPSK). It is found to have the same probability of error as QPSK and MSK techniques. It has a higher bandwidth efficiency in bits/sec/Hz than QPSK and MSK. It has 99.99 percent of its energy within the null bandwidth ·and its sidelobes are 63 db down from the main lobe. Intersymbol interference (ISI) was simulated on an HP 9845 computer and was shown to be smaller than the ISI in a QPSK or an MSK signal. Two different implementation schemes are presented.

Compounded codes are used for representing voice data in digital communication systems. This thesis addresses the use of the Mu-law companding algorithm in a system optimized for conferencing. A procedure for determining the degree of compression for a variable number of conferees and design equations for implementing a table-lookup scheme using read-only-memories are presented.