The goal of this bachelor thesis has been to investigate if the laboratory exercises in the courses digital filters and linear systems can be moved from matlab to a digital signal processor. The processor is a TMS320C6713 floating point processor mounted on a development board.   The original laboratories have been implemented and analyzed and some suggested changes have been presented for the digital filter laboration. For the laboration in linear systems, the exercise can be implemented as it is today. Furthermore, a transmultiplexer has been implemented and tested for real time execution.   Finally, an application programming interface has also been implemented, with common functions, used in the laboratories.

Sensor arrays are used in many applications where their ability to localize signal sources is essential. For many applications, it is necessary to estimate the direction-of-arrival (DOA) of target sources. Although there are many DOA estimation methods available, most of them are valid only for narrowband signals where time delay can be approximated as a phase shift. This thesis focuses on DOA estimation algorithms for wideband sources. Specifically, this thesis proposes the pruned fast beamformer which can reduce the number of computations of Delay-and-Sum (DS) beamforming by using a multi-resolution structure. For high resolution methods, signal subspace methods are required. Most of the subspace techniques for wideband signals decompose the received wideband signals into several bands of narrowband signals through bandpass filtering. Then, there are two different ways of processing decomposed signals. The incoherent methods process each band independently by a given narrowband method and average the results. The coherent methods attempt to modulate the signals in each band so that they can be combined coherently. In this thesis, a new DOA estimator, which is called TOPS, is developed to avoid disadvantages of both the incoherent and the coherent methods. The new method which can be categorized as a non-coherent method is tested and compared with other methods. It exhibits many desirable features for a number of applications where the sources are wideband such as acoustic direction finding.

Recent years have brought many new developments in the processing of speech and acoustic signals. Yet, despite this, the process of acquiring signals has gone largely unchanged. Adding spatial diversity to the repertoire of signal acquisition has long been known to offer advantages for processing signals further. The processing capabilities of mobile devices had not previously been able to handle the required computation to handle these previous streams of information. But current processing capabilities are such that the extra workload introduced by the addition of mutiple sensors on a mobile device are not over-burdensome. How these extra data streams can best be handled is still an open question. The present work deals with the examination of one type of spatial processing technique, known as beamforming. A microphone array test platform is constructed and verified through a number of beamforming agorithms. Issues related to speech acquisition through microphones arrays are discussed. The algorithms used for verification are presented in detail and compared to one another.

Thesis (Ph.D.)--University of Central Florida, 2008. / Adviser: Wasfy B. Mikhael. Includes bibliographical references (p. 152-166).

Thesis (PhD)--University of South Australia, 2001

Bibliography: p. 235-243. / xxvi, 243 p. : ill. ; 30 cm. / Title page, contents and abstract only. The complete thesis in print form is available from the University Library. / Concerned with the mathematical characterisation and adaptive processing of narrowband high frequency signals received by a very wide aperture antenna array after reflection from the ionosphere. / Thesis (Ph.D.)--University of Adelaide, Dept. of Electrical and Electronic Engineering, 2000

Thesis (Eng.D.) - University of Glasgow, 2007. / Eng.D. thesis submitted to the Faculty of Engineering, Department of Civil Engineering, University of Glasgow, 2007. Includes bibliographical references. Print version also available.

Thesis (Ph.D.)--McMaster University, 2003. / Advisor: James P. Reilly. Includes bibliographical reference (leaves 142-151). Also available via World Wide Web.

Increased usage of electronic devices and the fast development of microprocessors has increased the usage of digital filters ahead of analog filters. Digital filters offer great benefits over analog filters in that they are inexpensive, they can be reprogrammed easily and they open up whole new range of possibilities when it comes to Internet of things. This thesis describes development of a program that can sample music from the computer's microphone input, filter it inside the program with user built filters and reconstruct the music to the computer's headphone output meaning that the music can be played from the speakers. All of this is to happen in real time. The program is developed for students studying at the department of ``Signals and Systems" and the program is supposed the be one of the educational tools to make sense of signals and filtering. The program works well and filters the sound with satisfying results. It is easy to create filters and filter the signal. Since it is music that is filtered constructing perfect filters with minimum ripple, minimum or linear phase is quite difficult to achieve. The program could be improved by improving the user interface, making the environment more interactive and less difficult to construct good filters. Some improvements could also be made to the implementation; as of now the program might run a bit slow on startup on slower computers.

In this modern communication era, we are surrounded by unlimited electronic devices, the need to connect with everyone and everything increases dramatically. As the number of electronic devices increases, the amount of data to process increases and the need for higher data speeds occurs. After 1G and 2G, LTE (Long Term Evolution, 3G) come with the improvements in technology which allows reaching those new high data rates. Next comes the upgraded version of LTE called LTE-Advanced, which was launched to boost the speeds further. In this thesis, a 4G LTE environment has been created using (Graphic User Interface) in MATLAB. Many characteristics and parameters can be tuned such as type of modulation, number of UEs, type of channel, channel scenario, and some others to know how the system behaves and varied results outcome. Focus on the presence of a line of sight between the receiver and the transmitter helps to distinguish the Rayleigh and Rician scenario. In this thesis simulations on different channel models are simulated and various beamforming algorithms are tested to estimate that line of sight component (K-factor) and Error vector magnitude. The main aim of the thesis is to understand the communication channel behaviour in Static (Line of sight between transmitter and receiver) condition and High-Speed Train Condition along with EPA, EVA, ETU. The other aim of this report is to use the channel knowledge comprises of signal to noise ratio (SNR), bit error rate (BER) and error vector magnitude (EVM) helps to reduce the number of computations required while performing beamforming by varying the beam weight resolution.