No / In this paper the sensitivity of the decoupling matrix used for mutual coupling compensation in small size uniform circular arrays has been studied. The compensation matrix is calculated using the receiving mode technique for a 5-element uniform circular array and applied to two groups of direction finding algorithms, namely phase comparison-based (interferometry) and subspace-based algorithms. In the tracking application considered the receiver array is deployed on a car roof or aircraft, so the geometry of the platform influences the compensation results. In this work, the effect of different ground plane geometries in terms of the standard deviation of angular error for each estimation algorithm using simulation results is investigated. The results show that the calibration conditions used to determine the compensation matrix affect the AOA estimation accuracy.

In practice, there exist many error sources which distort the antenna array pattern. For example, elements of antenna arrays influence each other (known as mutual coupling), mismatches in cables and element positions affects the antenna radiation pattern and also unequal gain and phase characteristics of RF receiver distorts the received signal. These effects generally degrade the array performance. They cause an increase in sidelobe levels with an accompanying decrease in gain. Also, these errors limit the performance of direction finding (DF) algorithms. So, in order to have low sidelobe level, good performance in direction finding and beamforming, calibration is necessary. In the literature, there exist many algorithms proposed for the calibration of errors. Calibration method used in this thesis assumes that there is a linear transformation between ideal signal and measurements. Calibration matrix is formed by using measurements. In this work, we look for the adequate number of measurements for successful calibration. Performance of calibration method may depend on the angle interval from which measurements are taken. So, the width of the data collection angle interval is also studied. Moreover, in real life, measurements can be collected from equally or randomly spaced angles. Does it affect the performance of calibration? The answer of this question is also inspected in this thesis. Additionally, the performance of algorithm under noise is studied. Performance evaluation is done for both elevation and azimuth sectors. Simulations are carried out on MATLABTM and Ansoft HFSS software package.

In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further. / God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.

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