<p>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.</p><p> </p><p>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.</p><p> </p><p>Finally, an application programming interface has also been implemented, with common functions, used in the laboratories.</p><p> </p>

Bit-interleaved coded modulation with iterative decoding (BICM-ID)is a spectral efficient coded modulation technique to improve the performance of digital communication systems. It has been widely known that for fixed signal constellation, interleaver and error control code, signal mapping plays an important role in determining the error performance of a BICM-ID system. This thesis concentrates on signal mapping designs for BICM-ID systems. To this end, the distance criteria to find the best mapping in terms of the asymptotic performance are first analytically derived for different channel models. Such criteria are then used to find good mappings for various two-dimensional 8-ary constellations. The usefulness of the proposed mappings of 8-ary constellations is verified by both the error floor bound and simulation results. Moreover, new mappings are also proposed for BICM-ID systems employing the quadrature phase shift keying (QPSK) constellation. The new mappings are obtained by considering many QPSK symbols over a multiple symbol interval, which essentially creates hypercube constellations. Analytical and simulation results show that the use of the proposed mappings together with very simple convolutional codes can offer significant coding gains over the conventional BICM-ID systems for all the channel models considered. Such coding gains are achieved without any bandwidth nor power expansion and with a very small increase in the system complexity.

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.

Bit-interleaved coded modulation with iterative decoding (BICM-ID)is a spectral efficient coded modulation technique to improve the performance of digital communication systems. It has been widely known that for fixed signal constellation, interleaver and error control code, signal mapping plays an important role in determining the error performance of a BICM-ID system. This thesis concentrates on signal mapping designs for BICM-ID systems. To this end, the distance criteria to find the best mapping in terms of the asymptotic performance are first analytically derived for different channel models. Such criteria are then used to find good mappings for various two-dimensional 8-ary constellations. The usefulness of the proposed mappings of 8-ary constellations is verified by both the error floor bound and simulation results. Moreover, new mappings are also proposed for BICM-ID systems employing the quadrature phase shift keying (QPSK) constellation. The new mappings are obtained by considering many QPSK symbols over a multiple symbol interval, which essentially creates hypercube constellations. Analytical and simulation results show that the use of the proposed mappings together with very simple convolutional codes can offer significant coding gains over the conventional BICM-ID systems for all the channel models considered. Such coding gains are achieved without any bandwidth nor power expansion and with a very small increase in the system complexity.

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.

SUMO is a small ubiquitin-like modifier. The fluorescent fused SUMO (active for sumoylation) localized in the nucleus, while C-terminal truncated SUMO (inactive for sumoylation) diffused in the cytoplasm. Daxx is a SUMO target protein, locates predominantly in the nucleus. It has been identified as a component of the PODs. During extracellular stimulation, Daxx could be recruited to the cytoplasm with the existence of Ask1. Therefore, it is a shuttle protein. Daxx should contain nuclear localization signal (NLS) and nuclear export signal (NES) motifs. To identify the NES and NLS motifs on Daxx, Daxx were truncated into four segments. Several amino acids on the predicted NES and NLS motifs were mutated. Our results showed that the truncated Daxx fragments D1 (containing NES) and D4 (containing NLS2) could be translocated into nucleus independently. However, either NES or NLS2 mutants disrupted their translocation into nucleus. It indicated that both NES and NLS2 motif of Daxx were involved in the nuclear transport. Nevertheless the co-transfection of SUMOs and Daxx showed that the interactions between SUMO active form and Daxx mutants and between inactive SUMO and Daxx wild type rescued the nuclear transport function of Daxx mutants and inactive SUMO. Therefore, SUMO may play a role in the nuclear transport of Daxx by either sumoylation or interaction with Daxx in cytoplasm, and Daxx may recruit inactive SUMOs into nucleus by interaction.

The main advantage of wavelet transform relative to its Fourier analysis counterpart is its suitability to deal with transient signals. Furthermore, wavelet packet transform has very good frequency analytic ability with the result that it is developing in very fast speed and widespread researched and used in industry and academia. We study the characteristics of fiber optic sensors by applying wavelet transform. Hence, in this paper, the traditional Fourier analysis is taken as a basis, and the wavelet packet analysis is taken as a comparison. The major objects include: (1) calibration of hydrophones; (2) vibration measurement. In calibration of hydrophones, the experimental results show a 2.72 dB re V/£gPa inaccuracies and a 5.3 dB re V/£gPa standard deviation by Fourier analysis, but 0.5 dB inaccuracies and 1.6 dB re V/£gPa standard deviation by wavelet packet analysis. It shows that the wavelet packet analysis has better analytic ability than that of traditional Fourier analysis. In vibration measurement, we utilize FBG interferometers to measure stable vibration. The experimental results denote that wavelet packet analysis has excellent frequency analytic ability as Fourier analysis. Besides, in obtaining transient characteristic signals induced by falling stones, the results appear that wavelet packet analysis has better resolution and identification capability relative to Fourier analysis.

Techn. Universiẗat, Diss.--Dresden, 2006.

Thesis (Ph. D.)--Washington State University, December 2008. / Title from PDF title page (viewed on Sept. 23, 2008) "School of Electrical Engineering and Computer Science." Includes bibliographical references (p. 83-90).