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.

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.

Ultra wideband (UWB) signaling and its usability in positioning schemes has been discussed in this report. A description of UWB technology has been provided with a view on both the advantages and disadvantages involved. The main focus has been on Impulse Radio UWB (IR-UWB) since this is the most common way of emitting UWB signals. IR-UWB operates at a very large bandwidth at a low power. This is based on a technique that consists of emitting very short pulses (in the order of nanoseconds) at a very high rate. The result is low power consumption at the transmitter but an increased complexity at the receiver. The transmitter is based on the so-called Time Hopping UWB (TH-UWB) scheme while the receiver is a RAKE receiver with five branches. IR-UWB also provides good multipath properties, secure transmission, and accurate positioning whith the latter being the main focus of this report. Four positioning methods are presented with a view on finding which is the most suitable for UWB signaling. Received Signal Strength (RSS), Angle Of Arrival (AOA), Time Of Arrival (TOA) and Time Difference Of Arrival (TDOA) are all considered, and TDOA is found to be the most appropriate. Increasing the SNR or the effective bandwidth increases the accuracy of the time based positioning schemes. TDOA thus exploits the large bandwidth of UWB signals to achieve more accurate positioning in addition to synchronization advantages over TOA. The TDOA positioning scheme is tested under realistic conditions and the results are provided. A sensor network is simulated based on indications provided by WesternGeco. Each sensor consists of a transmitter and receiver which generate and receive signals transmitted over a channel modeled after the IEEE 802.15.SG3 channel model. It is shown that the transmitter power and sampling frequency, the distance between the nodes and the position of the target node all influence the accuracy of the positioning scheme. For a common sampling frequency of 55 GHz, power levels of -10 dBm, -7.5 dBm and -5 dBm are needed in order to achieve satisfactory positioning at distances of 8, 12, and 15 meters respectively. The need for choosing appropriate reference nodes for the cases when the target node is selected on the edges of the network is also pointed out.

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. / .

One of the major challenges in design of wireless networks is the use of the frequency spectrum. Numerous studies on spectrum utilization show that 70% of the allocated spectrum is in fact not utilized. This guides researchers to think about better ways for using the spectrum, giving rise to the concept of Cognitive Radio (CR). Maybe one of the main goals when designing a CR system is to achieve the best way of deciding when a user should be active and when not. In this thesis, the performance of Binary Power Allocation protocol is deeply analyzed under different conditions for a defined network. The main metric used is probability of outage, studying the behavior of the system for a wide range of values for different transmission parameters such as rate, outage probability constraints, protection radius, power ratio and maximum transmission power. All the studies will be performed with a network in which we have only one Primary User for each cell, communicating with a Base Station. This user will share this cell with N potential secondary users, randomly distributed in space, communicating with their respective secondary receivers, from which only M will be allowed to transmit according to the Binary Control Power protocol. In order to widely analyze the system and guide the reader to a better comprehension of its behavior, different considerations are taken. Firstly an ideal model with no error in the channel information acquisition and random switching “off” of the user is presented. Secondly, we will try to improve the behavior of the system by developing some different methods in the decision of dropping a user when it is resulting harmful for the primary user communication. Besides this, more realistic approaches of the channel state information are performed, including Log-normal and Gaussian error distributions. Methods and modifications used to reach the obtained analytical results are presented in detail, and these results are followed by simulation performances. Some results that do not accord with theoretical expectations are also presented and commented, in order to open further ways of developing and researching.

The interest in Wireless Sensor Networks is rapidly increasing due to their interesting advantages related to cost, coverage and network deployment. They are present in civil applications and in most scenarios depend upon the batteries which are the exclusive power source for the tiny sensor nodes. The energy consumption is an important issue for research, and many interesting projects have been developed in several areas. They focus on topology topics, Medium Access Control or physical issues. Many projects aim at the physical layer where the node's power consumption is optimized through scaling the modulation scheme used in node communications. Results show that an optimal modulation scheme can lead to the minimum power consumption over the whole wireless sensor network. A usual simplification in research is to target individual paths and not take into account the whole network. However nodes may be part of several paths, and therefore nodes closer to the sinks may consume higher amounts of energy. This fact is the chief motivation of our research, where modulation scaling over the nodes with more energy is performed in order to increase the lifetime of the nodes having lower energy reserves. Simulation results showed typical values of path lifetime expectancy of 50 to 120 percent higher than comparable power-aware methods.

This report is a continuation of the student project "Evaluation of TrinnovOptimizer audio reproduction system". It will further investigate theproperties and function of the Trinnov Optimizer, a correction system foraudio reproduction systems. During the student project measurements wereperformed in an anechoic lab to provide information on the functionality andabilities of the Trinnov Optimizer. Massive amounts of data were recorded,and that has also been the foundation of this report. The new work that hasbeen done is by interpreting these results through the use of Matlab. The Optimizer by Trinnov [9 ] is a standalone system for reproductionof audio over a single or multiple loudspeaker setup. It is designed tocorrect frequency and phase response in addition to correcting loudspeakerplacements and cancel simple early re?ections in a multiple loudspeakersetup. The purpose of further investigating this issue was to understandmore about the sound?eld produced around the listening position, and togive more detailed results on the changes in the sound?eld after correction.Importance of correcting the system not only in the listening position, butalso in the surrounding area, is obvious because there is often more than onelistener. This report gives further insight in physical measurements ratherthan subjective statements, on the performance of a room and loudspeakercorrection device. WinMLS has been used to measure the system with single, and multiplemicrophone setups. Some results from the earlier student project are alsoin this report to verify measurement methods, and to show correspondancebetween the di?erent measuring systems. Therefore some of the data havebeen compared to the Trinnov Optimizer's own measurements and appear similar in this report. Some errors found in the initial report, the results from the phase response measurements, have also been corrected. Multiple loudspeakers in a 5.0 setup have been measured with 5 microphones on a rotating boom to measure the soundpressure over an area around the listening position. This allowed the e?ect of simple re?ections cancellation, and the ability to generate virtual sources to be investigated. For the speci?c cases that were investigated in this report, the Optimizer showed the following: ? Frequency and phase response will in every situation be optimized to the extent of the Optimizers algorithms. ? Every case shows improvement in the frequency and phase response over the whole measured area. ? Direct frontal re?ections was deconvolved up to 300Hz over the whole measured area with a radius of 56cm. ? A re?ection from the side was deconvolved roughly up to 200Hz for microphones 1 through 3, up to a radius of 31.25cm, and up to 100Hz for microphones 4 and 5. ? The ability to create virtual sources corresponds fairly to the theoretical expectations. The video sequences that were developed give an interesting new angle on the problems that were investigated. Other than looking at plots of di?erent angles which is di?cult and time consuming, the videos showed an intuitive perspective that enlightened the same issues as the common presented data of frequency and phase response measurements.

One of the fundamentals concerns in wireless communications with battery operated terminals is the battery life. Basically there are two ways of reducing power consumption: the algorithms should be simple and efficiently implemented (at least in the wireless terminals), and the transmit power should be limited. In this document is considered discrete time, progressive signal transmission with feedback [ramstad]. For forward Gaussian channel, with an ideal feedback channel, the system performs according to OPTA (Optimal Performance Theoretically Attainable[berger]). In this case, with substantial bandwidth expansion through multiple retransmissions, the power can be lowered to a theoretical minimum. In the case of a non-ideal return channel the results are limited by the feedback channel's signal-to-noise ratio. Going one step forward, a more realistic view of the channel will consider fading due to multiple reflections, especially in indoors scenarios. In this thesis it is discussed how to model the channel fading and how to simulate it from different probability distributions. After, some solutions to avoid, or at least reduce, all the undesirable effects caused by the fading will be proposed. In these solutions, the fading characteristics (power and dynamic range) and the application requirements will play a vary important role in the final system design. Finally, a realistic signal will be tried to be sent in a realistic scenario. This will be audio transmission over fading channels. Then, the results will be compared in general terms to a similar equipment such as generic wireless microphone system.

Krysstale mellom tvinnede parkabler er blitt den dominerende kilden for interferens i dagens DSL (Digital Subscriber Lines) systemer. I eksisterende standarder er det satt maksimalverdier for utsendt effekttetthet basert på estimat av verste tilfelle interferens for alle par-til-par kombinasjoner av krysstale med følgelig begrensninger i overføringskapasitet. Videre blir effekt statisk tildelt over hele den tilgjengelige båndbredden uten hensyn til frekvensavhengig attenuasjon og interferens i metoder basert på statisk frekvensbruk (SSM - Static Spectrum Management). Dette arbeidet tar for seg utviklingen av autonome algoritmer for dynamisk frekvensbruk (DSM – Dynamic Spectrum Management) innen trådbasert kommunikasjon over det eksisterende kobbernettet. DSM åpner for effekttildeling basert på målinger av kanalvariasjoner i frekvensbåndet og terminaler kan slik begrense sin egen utsendte effekttetthet dersom ønsket overføringsrate allerede er oppnådd. En slik dynamisk allokering av spektrum gjør det mulig å prioritere frekvensbånd med høye signal-til-støy forhold (SNR) samtidig som sterkt forstyrrede delbånd får kraftig reduksjon av tildelt effekt eller blir fullstendig slått av. Ulike algoritmer ble implementert og analysert, med den velkjente water-filling algoritmen spesielt sentral i beregningene, som gav svært gode resultater med hensyn til eksisterende effektallokering basert på SSM.

This work is a study about source localization methods, more precisely, about beamforming approaches. The necessary background theory is provided first, and then, further developed to explain the basis of each approach. The studied problem consists in an array of sensors in which the signal to process is impinging. Several examples of inciding signals are provided in order to compare the performance of the methods. The goal of the approaches is to find the Incident Signal Power and the Direction Of Arrival of the Signal (or Signals) Of Interest. With these information, the source can be located in angle and range. After the study, the conclusions will show which methods to chose depending on the application pursued. Finally, some ideas or guidelines about future investigation on the field, will be given.