<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>

<p>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.</p>

<p>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.</p>

<p>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.</p>

<p>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.</p>