Thesis (M. Eng.)--Carleton University, 1997. / Includes bibliographical references. Also available in electronic format on the Internet.

Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2010. / Committee Chair: Altunbasak, Yucel; Committee Co-Chair: Zhou, G. Tong; Committee Member: Al-Regib, Ghassan; Committee Member: Kenney, James Stevenson; Committee Member: Ma, Xiaoli; Committee Member: Pan, Ronghua. Part of the SMARTech Electronic Thesis and Dissertation Collection.

Thesis (M.S.)--Brigham Young University. Dept. of Computer Science, 2006. / Includes bibliographical references (p. 109-113).

Thesis (M.S.)--Ohio University, August, 2004. / Title from PDF t.p. Includes bibliographical references (p. 81-83)

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).

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.

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.

This thesis investigates the behavior of two digital communication systems based on Moving-Average Matched Filters (MAMF). In general, matched filters are instrumental in detecting signals corrupted by noise as they are designed to maximize the probability of detection of the transmitted signals. The MAMF represents a subset of the class of matched filters. The two communication systems under investigation are the classical MAMF system and one of its modifications, the proposed MAMF system. In the traditional system the N-dimensional signal vector, which encodes the bit to be communicated, remains fixed throughout the whole communication process ( transmission and reception). In the proposed system the encoding N-dimensional signal vector is composed of K linearly independent basis vectors spanning a signal vector subspace of dimension M (= N /K). By combining these basis vectors in the receiver, any vector in the signal vector subspace can be formed in order to maximize the Output Signal-to-Noise Ratio (OSNR). The relative measure of comparison for the two systems is the Signal-to-Noise Ratio Improvement (SNRI). The SNRI is the ratio of the OSNR, which is measured at the output of the receiver, to the Input Signal-to-Noise Ratio (ISNR), which is measured at the input of the receiver. Since the ISNR is fixed for a particular transmitted signal vector and noise characteristics, an attempt is made to maximize the SNRI by maximizing the OSNR. / Master of Science

<p>This thesis presents the lessons learned from building an IEEE 802.11 wireless mesh network (WMN) test-bed. Each network node consists of a Linux processor with multiple IEEE 802.11b/g transceivers operating in the 2.4 GHz band. Each transceiver consists of a medium access control (MAC) and base-band processor (BBP) in addition to a radio. A device driver was modified to control some of the key transceiver functions. The test-bed's Wi-Fi interfaces can be programmed to implement any mesh communication topology. All Wi-Fi interfaces use omni-directional antennas and the IEEE 802.11b operation mode.</p> <p>The test-bed design is easily extendable to incorporate newer Wi-Fi technologies. Measurements of co-channel interference in each Wi-Fi channel including received signal strength (RSS) and signal-to-interference-and-noise ratio (SINR) are presented. The AutoMin algorithm was developed in order to use the captured physical layer (PHY) metrics to avoid Wi-Fi congestion during test-bed operation. A comparison of a software-based spectrum analyzer to a commercial one is described. Key Wi-Fi functions in the Ralink driver source code are explored in depth. The compliance of the Ralink chip-set to the IEEE 802.11b spectral mask was verified. The maximum driver-induced retuning rate for the popular Ralink radio was found experimentally. This data can be used to optimize the performance of IEEE 802.11 WMNs.</p> / Master of Applied Science (MASc)