Grunnlaget for det arbeidet som er utført i denne rapporten er påstanden om LoranCsrobusthet som navigasjonssystem. Robustheten til systemet måles i systemets egen evnetil å motstå interferens (absolutt robusthet) og hvor robust LoranC er i forhold til andre navigasjonssystemer(relativ robusthet). Fokuset i denne rapporten er å gi en beskrivelse av navigasjonssystemet LoranC, hvilke faktorer som påvirker signalet langs utbredelsesveien og hva som skal til for å interferere med disse signalene. For å avdekke hva som kan gjøres for å interferer med LoranC signalene er det utført praktiske målinger/jammetest. De teoretiske beregningene sammen med observasjonene i det praktiske forsøket skal danne grunnlaget for et hensiktsmessig oppsett av en mobil LoranC jammerenhet.Ønsker man å påvirke en mottakers posisjonsangivelse er det viktig at man kjennertil virkemåten til systemet og prinsippene brukt i mottakerene. Kapittel 4.2 tar for segde teoretiske aspektene ved LoranC systemet og kap 4.3 omhandler virkemåten til enLoranC mottaker. Søken etter å finne sårbarheten til systemet startes med å se på de potensielle feilkildene som systemet må ta høyde for under normal drift. LoranC signalene påvirkes av underlaget(den elektriske ledeevnen), topografien, årstiden og de aktuelle atmosfæriske forholdene. Dette er degraderende faktorer som LoranC-systemet må ta med i beregningene for å oppnå systemkravene til posisjonsnøyaktighet. For å gi en bedre forståelse for hva LoranC signalene ved normal drift påvirkes av vil kap.4 gi en gjennomgang av utbredelse av LoranC-signaler og hvordan den atmosfæriske støyen påvirker disse signalene.Videre gir kap 4.8 en introduksjon av generelle signaltyper og modulasjoner som kanbenyttes til å interferere med LoranC-signalene. Dette er signaler som videre skal benyttes som jamme/narre signaler i de praktiske målingene.Grensesnittet mellom det elektriske interferenssignalet(jamming/spoofing) og utbredelsesmediumet(eteren) er antennen, og på grunn av at LoranC er et lavfrekvent navigasjonssystem vil signalenes lange bølgelengden skape utfordringer ved antennedesignet.I denne rapporten vi dette gjøre seg mest gjeldende ved at størrelsen til LoranC senderantennen begrenses av kravet til mobilitet hos jammeenheten, mens god antenneeffektivitet krever en fysisk stor antenne. Kap 4.9 gir en oversikt over disse utfordringene og viser beregninger av en antenne som kan benyttes i et slikt mobilt jammeoppsett.For beregning av feltstyrkeverdiene til LoranC-signalene og den atmosfæriske støyen vedmålestasjonen ble det utviklet og brukt et Excel-program med utgangspunkt i drøftet bølge- og støy-teori. Kapittel 5 gir en gjennomgang av dette Excel-programmet. Beregningene i dette programmet danner grunnlaget for hva som kreves av en jammeenhetsom skal kunne interferere med LoranC signaler fra tre valgte sendere. Hva som krevesav en slik jammerenhet er presentert i dette kapittel.I kapittel 6 beskriver de praktiske målingene som ble utført på Frøya. For å avdekke hvasom er den mest effektive formen for interferens med LoranC-signaler ble interferensmålingene utført ved bruk av to typer antenner, forskjellige signaltyper/modulasjoner og varierende signalstyrke/avstand. Videre ble de teoretiske signalberegninger for LoranCsignalet og den atmosfæriske støyen sammenlignet med signal-verdier observert under jammemålingene. Egenskapene og ytelsen til jammeantennene og posisjonsplottet fra LoranC mottakeren ble også observert og benyttet til å fastslå effektiviteten av interferensen med LoranC-signalene og jamme-effektivitet. Denne rapporten avsluttes med at de teoretiske resultatene i kombinasjon med de praktiske undersøkelsene gir en bedømmelse av LoranC sin robusthet samt et oppsett av et mobilt LoranC-jammesystem.

<p>This Master's thesis reports the verification planning and verification process of a Verilog RTL model. Modern verification techniques like constrained randomization, assertions, functional coverage analysis and object orientation are demonstrated on an AES RTL model. The work of this thesis was naturally divided in three phases: First, a phase of literature studies to get to know the basics of verification. Second, the creation of a verification plan for the selected module. Third, implementation of the testbench, and simulation tasks. The verification plan created states the goals for the simulation. It also states plans for details about the testbench, like architecture, stimuli generation, random- ization, assertions, and coverage collection. The implementation was done using the SystemVerilog language. The testbench was simulated using the Synopsys VCS ver- ification software. During simulation, coverage metrics were analyzed to track the progress and completeness of the simulation. Assertions were analyzed to check for errors in the behavior during simulation. The analysis carried out revealed high code coverage for the simulations, and no major errors in the verified module.</p>

<p>In this thesis we take a closer look at wireless sensor networks and source coding. A necessary condition for this work to have any meaning is that the sensors in the network are spatially co-located and that there is correlation between the data the sensors observe. When there is correlation, there is redundancy in the information communicated that can be removed by source coding techniques. This can be done by emph{distributed source coding}. Slepian and Wolf showed theoretically that there is no rate loss no matter if the sensors are communicating. cite{slepian73} Wyner and Ziv expanded this from the lossless case of Slepian and Wolf to apply to lossy source coding. cite{wyner76} Pradhan and Ramchandran found a practical implementation for the theory of Slepian-Wolf and Wyner-Ziv based on channel coding principles. cite{pradhan03} This can be done because the correlation between any two sources can be modelled as a channel with an error probability. We build our work on their ideas. The channel coding technique we have found most advantageous for this scheme is emph{Low Density Parity-Check} coding. LDPC coding is the most advanced form of linear block coding up to date. It is represented by a sparse parity-check matrix. While LDPC coding in the traditional sense is used for bandwidth expansion of the source to protect it from channel errors, it is used for bandwidth compression, or rate reduction, in the distributed sense. The distributed LDPC scheme is used on medical ECG data as an example. Due to lack of time and the comprehensive task, the adpapted message-passing decoding algorithm needed to fulfill the implementation could not be finished. We have illustrated the distributed encoder system with a $(7,4)$-Hamming code to give an example. The performance of this system is not good enough for any practical use, but will function as a guideline for possible future work in the area.</p>

<p>In the fall 2006 NTNU (The Norwegian University and Science and Technology) initiated a strategic project in cooperations with SINTEF where the aim is to gain more knowledge about underwater acoustic communications. This study is a part of this project and focuses on a system for underwater communication. A orthogonal frequency division multiplexing (OFDM) system using differential quadrature phase shift keying (DQPSK) has been defined and implemented in MATLAB. The system has been characterized through thorough simulations and testing. Initial measurements has also been carried out in order to test the developed system on a real underwater acoustic channel and the results have been analysed.</p>

<p>Among several different damaging mechanisms, oxidative stress is found to play an important role in noise-induced hearing loss (NIHL). This is supported by both findings of oxidative damage after noise exposure, and the fact that upregulation of antioxidant defenses seem to reduce the ears susceptibility to noise. Oxidative stress mechanisms could help explain several of the characteristics of NIHL, and we therefore believe that it would be advantageous to estimate noise-induced hearing impairment on the basis of these, rather than the prevailing energy based methods. In this thesis we have tried to model progress of NIHL using diffusion principles, under the assumption that accumulation of reactive oxygen species (ROS) is the cause of hearing impairment. Production, and the subsequent accumulation, of ROS in a group of outer hair cells (OHCs) is assessed by different implementations of sound pressure as in-parameter, and the ROS concentration is used in estimation of noise-induced threshold shift. The amount of stress experienced by the ear is implemented as a summation of ROS concentration with different exponents of power. Measured asymptotic threshold shift (ATS) values are used as a calibrator for the development of threshold shifts. Additionally the results are evaluated in comparison to the standards developed by the International Organization for Standardization (ISO) and the American Occupational Safety and Health Administration (OSHA). Results indicate that ROS production is not directly proportional to the sound pressure, rather anaccelerated formation and accumulation for increasing sound pressure levels (SPLs). Indications are also that the correlation between concentration of ROS and either temporary threshold shift (TTS) and/or permanent threshold shift (PTS) is more complex than our assumption. Because our model is based on diffusion principles we get the same tendency of noise-induced hearing loss development as experimentally measured TTS development. It also takes into account the potentially damaging mechanisms which occur during recovery after exposure, and has the ability to use TTS data for calibration. We therefore suggest that modeling of ROS accumulation in the hair cells could be used advantageously to estimate noise-induced hearing loss.</p> / .

<p>Knowledge about the physcical parameters of the seafloor is often important information. This master’s thesis looks at seismic inversion to find these parameters. The choice of forward model is highly emphasised. A seismic inversion has a number of variables which can be changed and altered to obtain a good result. The forward model will have a big impact on the results of the inversion. Both the time spent on the inversion, and which parameters the inversion will be best suited to estimate will be determined by the choice of forward model. An inversion code written in Matlab by Fredrik Helland is used. It uses genethic algorithms as optimization, and OSIRIS as forward model. This code is expanded to deal with several forward models and seafloor geometries. Testing of the inversion code shows that all the forward models serves different perposes. The ray tracing model is still at a consept level, but should be usable in the future when it runs a bit faster and can deal with more than 3 layers. The dispersion method and the wave number integration method both work well and the results show that using a combination of them might be the best choice if all the geoacoustic parameters of the seafloor is sought.</p>

<p>The paper presents experiments made in search of answers to two principal questions: 1. Can one single musician be made to sound like several musicians playing together? 2. In a music ensemble, where one of its constituents has a distinctive spectrum; how do the deviant spectral components influence a listener’s ability of localising the source? In the first part of the experiment, a flute ensemble was attempted simulated. Based on a recoring of one flute playing a short piece, the flute was multiplied into a quintet. On the way, several properties were manipulated in an attempt to make the quintet sound like a real quintet; timing, spectrum, intensity, and phase. In the second part, one flute in a quintet was subject to a spectral tilt, i.e. high frequency components were boosted while low frequency components were diminished. A test panel was engaged to help evaluating the questions. First, the panel compared the simulated quintet to a reference quintet, trying to identify the simulation from the reference. Subsequently, listening to a reference quintet, the panel tried to localise the one flute which had undergone a spectral tilt. A musical piece was played 5 times; first, one of the flutes was moderately tilted, then the tilt’s magnitude was increased for every run until eventually being noticeable. For each run, the test panel was asked to indicate the tilted flute, or a random flute if none appeared tilted to them. The majority of the test panel did not manage to tell the simulated quintet from the reference. However, the reference may have been imperfect, and the simulation process somewhat affects sound quality. When it comes to localisation, a rather excessive tilt was necessary for the test panel to be able to localise it - even though more moderate tilts were clearly audible.</p>

<p>Bunn absorbent ble designet til å matche overliggende mediet.</p>

<p>After the planning phase autumn 2006, the work with the student satellite project evolved into sub-system design and prototyping. The work presented in this report considers a proposal for a VHF radio system intended for a small student satellite. The design process started on scratch, not looking much at earlier ncube designs, almost no documentation is to be found about actual construction and final measurements. Three design concepts where developed, one featuring an integrated transceiver, one as a self-designed FSK radio and the last one uses a GMSK-modem to solve modulation and de-modulation issues. As the design was chosen and the work of selecting components commenced, it became clear the chosen design would become not unlike the receiver proposed for ncube. The reason for this is component availability, especially the SA606 IF-sub-system and the GMSK-modem. During test and measurement, a few issues were discovered. The proposed low noise amplifiers seems to be a dead end for this frequencies, and alternatives must be found. The layout for the SA606 is improved and seems to function as required. Since the chosen layout is quite similar to the previous ncube 145 MHz receiver, it shows that the components selected for this designs are a good solution. However, the design is so extensive more work is required before a prototype is ready. It can be questioned if the first design proposal would have been less extensive and could have lead to a finished prototype withing the assigned time frame. Anyway, link budgets and power estimates shows that it is possible to build such a system within the defined limits.</p>

<p>Compensation of loudspeaker nonlinearities is investigated. A compensation system, based a loudspeaker model (a computer simulation of the real loudspeaker), is first simulated in matlab and later implemented on DSP for realtime testing. So far it is a pure feedforward system, meaning that no feedback measurement of the loudspeaker is used. Loudspeaker parameters are drifting due to temperature and aging. This reduces the performance of the compensation. To fulfil the system, an online tracking of the loudspeaker linear parameters is needed (also known as parameter identification). Previous investigations (done by Andrew Bright and also Bo R. Pedersen) shows that the loudspeaker linear parameters can be found by calculations based on measurements of the loudspeakers current. This is a subject for further work. Without the parameter identification, the compensation system is briefly tested, with the loudspeaker diaphragm excursion as output measure. The loudspeaker output and the output of the loudspeaker model are both monitored, and the loudspeaker model is manually adjusted to fit the real loudspeaker. This is done by realtime tuning on DSP. The system seems to work for some input frequencies and do not work for others.</p>