<p>Wireless ad hoc networks have a number of advantages over traditional, infrastructure-based networks. Robustness and easy deployment are two of the main advantages. However, the distributed nature of such networks raises a number of design challenges, especially when energy-efficiency and QoS requirements are to be taken into consideration. These challenges can only be met by allowing closer cooperation and mutual adaptation between the protocol layers, referred to as a cross-layer design paradigm. In energy-constrained wireless ad hoc networks, each node can only transmit to a limited number of other nodes directly. Hence, in order to reach distant destinations, intermediate nodes must relay the traffic of their peer nodes, resulting in multihop routes. The total energy consumption associated with a end-to-end transmission over such a route can be significantly reduced if the nodes are correctly configured. A cross-layer, optimization scheme, based on adaptive modulation and power control, is proposed in this thesis. The optimization scheme assumes that an existing route has been found, and allows QoS requirements in terms of end-to-end bit error rate and delay. Both transmission and circuit energy consumption is taken into consideration. By jointly optimizing all nodes throughout the route, the total energy consumption can be reduced by more than 50%, compared to a fixed-rate system. The adaptive system also exhibits superior capabilities to meet stringent QoS requirements. Results for both continuous and discrete rate adaptation is produced, and it is found that discrete adaptation causes only a small performance degradation, compared to the optimal, continuous case. Simulations also show that the system is vulnerable to inaccurate link state information. Finally, the effects of maximum-rate limitation and ignoring the circuit power consumption is investigated.</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>Cellular networks are widely deployed for wireless communication, and as the number of users of these networks increase, so does the need for higher spectral efficiency. Clever measures have to be taken in order to increase throughput for wireless networks because of the scarcity of radio resources. Ever higher rates are demanded, but we also want to conserve a fair distribution of the available resources. Therefore, we consider the problem of joint power allocation and user scheduling, while achieving a desired level of fairness in wireless cellular systems. Dynamic resource allocation is employed for the full reuse networks simulated, in order to cope with inter-cell interference and to optimize spectrum efficiency. Binary power allocation is implemented and compared to the performance without power control, for minimum transmit power levels equal to 0 and greater than 0. We show that binary power control with individual power levels for each cell is optimal for two-cell networks. We also present an extension to the proportional fair scheduling for multi-cell networks, and analyze its performance for different cell sizes and time windows. Finally, we highlight the equality between multi-cell, multi-user and multi-carrier proportional fair scheduling. Simulation results show how power control and user scheduling increase throughput, reduce power consumption and achieve a desired level of fairness. Hence, we can obtain considerable gains for the network throughput through adaptive power allocation and multiuser diversity.</p>

<p>Wireless Interface for Sensors and Actuators (WISA) is ABB's proprietary wireless protocol for industrial automation on the factory floor. It operates in the 2.4GHz ISM band. Wireless Local Area Networks (WLANs), which typically occupy a fixed portion of the same 2.4GHz ISM band, are becoming more and more common on the factory floor. This raises a question of co-existence and how the performance of traffic over WLAN is affected when interfered by WISA. This report is a result of the development of a software tool and assembly of hardware that can aid the future testing of the effect WISA has on nearby WLANs. Together with the explanation of the usage of this software tool, this report will also investigate different arrangements of hardware components that are used to demonstrate and test the functionality of this new software tool. The software tool and the hardware components enable the measurement of important traffic metrics between two computers that communicate over a WLAN. The hardware components include a WISA Base Station (BS) that is configurable through the software tool and is used to cause interference on the WLAN.</p>

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

<p>Wireless Interface for Sensors and Actuators (WISA) is ABB’s proprietary wireless protocol for industrial automation. It operates in the 2.4 GHz ISM band, as do nearly allWireless LAN systems. WISA does frequency-hopping (FH) over most of the ISM band, but has currently no means of avoiding parts of the band occupied by other wireless systems. The objective of the diploma project was to create a Matlab based simulation tool that can (i) analyze cross-correlation between FH sequences in two closely spaced WISA cells, and (ii) generate new FH sequences which avoid a user-selectable portion of the frequency band. New frequency-hopping sequences were designed using Galois field computations for creating periodic sequences with minimum correlation. The developed Matlab simulation module did indeed meet the objectives. However the algorithm for subband-allocation is not optimal and will for some cases not give maximum utilization of the available frequency band. Analysis of the existing FH algorithm confirmed that some sequence pairs are non-ideal in the sense that their inevitable frequency collisions are not spread evenly over all relative shifts between the sequences, but concentrated to a few of these shifts. It was also pointed out that not all cell ids met the desired requirement of large separation between transmissions occurring on consecutive frames. Analysis of the new FH sequences, which avoid a user-selected portion of the frequency band, showed that these had many of the same properties as the existing algorithm. It was possible to find sequence pairs with low correlation and thus allow multiple cells to operate in the same radio space.</p>

<p>Many network-based multimedia applications transmit real-time media over unreliable networks, i.e. data may be lost or corrupted on its route from sender to receiver. Such errors may cause a severe degradation in perceptual quality. It is important to apply techniques that improve the robustness against errors, in order to ensure that the receiver is able to playback the media with the best attainable quality. Today, most ER schemes for video employ proactive error resilient encoding. These schemes add redundant information into the encoded video stream in order to increase the robustness against potential errors. Because of this, most proactive schemes suffer from a significant reduction of the coding efficiency. Another approach is to adjust the encoder operations based on feedback information from the decoder, e.g. to repair corrupted regions based on reports of lost data. Feedback-based ER schemes normally improves the coding efficiency compared with proactive schemes. Moreover, they adjust rapidly to time-varying network conditions. The objective of this thesis is to develop and evaluate a feedback-based ER scheme conforming to the H.264/AVC standard and applicable for real-time low-delay video applications. The scheme is referred to as FBIR. The performance of FBIR will be compared with an existing proactive ER scheme, known as IPLR. Special attention is given to the applied feedback mechanism, RTP/AVPF. RTP/AVPF is a new (2006) feedback protocol. Basically, it specifies two modifications/additions to the RTCP: First, it modifies the timing algorithm to enable early feedback, while not exceeding the RTCP bandwidth constraint. Second, new RTCP message types are defined, which provides information useful for error control purposes. FBIR employs RTP/AVPF to provide timely feedback of lost packets from the decoder to the encoder. Upon reception of this feedback, the encoder use a fast error tracking algorithm to locate the erroneous regions. Finally, the regions that are assumed to be visually corrupted after decoding are intra refreshed. IPLR is an ER scheme developed for use in a commercial video communication system. It applies a motion-based intra refresh routine. The comparison is carried out by online simulations with various network environments (0, 1, 3 and 5% loss rate; 50 and 200 ms latency), bit rates (64, 144 and 384 kbit/s) and video sequences. First, the video is encoded and transmitted in real-time to the decoder via a network emulator. This emulator generates the desired network characteristics. The receiver decodes the video in real-time and transmits feedback information back to the encoder. The encoder adjusts its encoding process according to this feedback. The H.264/AVC reference software is modified and used as codec. Finally, objective quality measures are obtained by calculating the PSNR of the decoded videos. In addition, some visual inspection is performed. Isolated measures on the RTP/AVPF transmission algorithm are also performed. These show that RTP/AVPF is able to provide timely feedback for error control purposes for a great number of applications and network environments. However, the experienced feedback delay may be increased by numerous factors, e.g. the network latency, the packet loss rate, the session bandwidth, and the number of receivers. This may decrease the performance of ER schemes utilizing RTP/AVPF. RTP/AVPF is fairly easy to implement since it only modifies the RTCP timing algorithm and adds new RTCP message types. RTP/AVPF may be used in combination with other standards in order to extend the available feedback information. Hence, RTP/AVPF enables timely feedback for use in a wide range of multimedia applications. The PSNR measurements show that FBIR always obtains higher objective quality than IPLR for error free transmissions. This does not, however, necessarily affect the perceptual quality if the bit rate is high. FBIR achieves higher PSNR in other situations as well, such as for very low loss rates, low or medium bit rates, and for sequences with high or medium motion activity. Conversely, IPLR performs better for low motion sequences encoded at high bit rates when the loss rate exceeds a certain threshold, typically about 1%. It is also shown that the performance of FBIR may be reduced if the network latency increases. Visually, the main difference between the two schemes is that FBIR recovers all corrupted regions at one instant, while IPLR performs a gradual refresh. The average time before recovery is somewhat shorter for IPLR. The differences between FBIR and IPLR are mainly caused by two factors. First, using FBIR results in less intra coding and thus better coding efficiency. Second, the FBIR scheme does not repair errors until the encoder receives the feedback. Usually, this happens after IPLR has repaired most of the corrupted region. In short, one can say that FBIR provides medium error robustness and high coding efficiency, in contrast to IPLR's high robustness and low coding efficiency. While FBIR's performance may be reduced by network characteristics such as increased latency, IPLR is unaffected by these factors. For error free transmissions, FBIR does not significantly reduce the coding gain compared with a non-robust encoding scheme. Still, it provides a good robustness against corruption in error-prone networks. Thus, all real-time video systems that benefit from immediate feedback should strongly consider to employ FBIR or similar feedback-based ER schemes.</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>