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  • 301. Nordberg, Jörgen
    et al.
    Dam, Hai Huyen
    Nordholm, Sven
    A Blind Closed Loop Subband Channel Equalizer2004Conference paper (Refereed)
    Abstract [en]

    The wireless communication channel are characterized by its multipath richness, specially in the unlicensed 60 GHz band where many of the modern and future wireless communication system will work. To reverse the effects of even a moderate delay spread, in the linear equalization case, will require an estimation of high order filters. Thus, there will be a demand for low complexity high quality receivers that should be able to work during sever multipath conditions. In this paper, a new blind closed loop subband equalizer is proposed to reduce the computational complexity of the equalizer and also to improve the channel tracking capability. Simulation results show a significantly improvement, 12 times, of the convergence rate without any significant performance loss.

  • 302. Nordebo, Sven
    et al.
    Mohammed, Abbas
    A Semi-definite Programming Approach to Spatial Decorrelation of Independently Polarized Signals2007In: Wireless Communications & Mobile Computing, ISSN 1530-8669, E-ISSN 1530-8677, Vol. 7, no 1, p. 91-101Article in journal (Refereed)
    Abstract [en]

    A MIMO channel spatial decorrelation scheme based on semi-definite programming is introduced. As a particular application example, the paper addresses the potential gain of using multiple antennas and MIMO-OFDM techniques in order to increase the bandwidth efficiency in satellite communication systems. In particular, we consider the increase in channel capacity that is possible by exploiting satellite and polarization diversity. A fundamental case is studied with three satellite branches, and where each transmit/receive antenna unit consists of six elemental electric and magnetic dipoles yielding six distinguishable parallel polarization channels per frequency. The numerical examples show that capacity increases linearly on a logarithmic signal-to-noise ratio scale where the constant of proportionality is the number of active parallel channels. In this respect, the simultaneous use of triple electric and triple magnetic dipoles has the potential to triple the capacity of an antenna system based on antenna units of single dipoles. Copyright (c) 2006 John Wiley & Sons, Ltd.

  • 303. Nordholm, Sven
    et al.
    Low, Siow Yong
    Claesson, Ingvar
    Yiu, Ka Fai Cedric
    Non-uniform Optimal Subband Beamforming: An Evaluation on Real Acoustic Measurements2008Conference paper (Refereed)
    Abstract [en]

    This paper presents an evaluation of a subband beamforming scheme for speech enhancement and acoustic echo suppression applications, such as hands-free telephony, internet telephony and video conferencing. The aim of the paper is to study the impact of the different types of subband filter banks on the speech enhancement performance. Results show that the octave band filter banks provide an effective means to give an overall efficient processing by using a fewer free filter parameters while providing a similar performance in terms of Signal to Noise Interference Ratio (SNIR) as compared to the uniform case. Whilst the critical band filter bank uses the most subbands, it provides the best performance both objectively and subjectively. This study provides insight to the design of filter bank for future hardware implementations. © 2008 IEEE.

  • 304.
    Olmedilla, Carlos Hernandez Matas and Diana Gomez
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Image Theory Applied to virtual Microphones2008Independent thesis Basic level (degree of Bachelor)Student thesis
    Abstract [en]

    Virtual sensing is the art of measuring a quantity at a certain spatial position without having a physical sensor placed at that exact same position. In practical applications, this technique is sometimes very useful, when it is not feasible to put the sensor at the position where the physical quantity should be measured. For example, a virtual microphone could be considered in the following scenario: The sound level, preliminary school environment, should be studied at the different students head positions. In order to get statistical relevant data, the measurement has to be carried out over a period of time. However, it is not a feasible approach to use a large number of microphones hanging from the ceiling at the students head level, over a longer period of time for obvious reasons. One thoughtful solution, in such a situation is to put microphones at the walls and close to the ceiling and using the virtual technique to calculate the noise level at the students head positions based on the measure data. An advantage in such large setup is that it often requires less physical sensors than the number of virtual measurement positions and at the same time features a better validation and sanity check of the data. The most popular application areas for virtual microphones are in active noise control (ANC) or active structural acoustic control (ASAC), where the aim is to move the zone of quietness away from the physical error microphones to the desired location of maximum attenuation. However, there is an important difference between virtual sensing for active control and virtual sensing for a pure measurement situation concerning the delays introduced by virtual sensing technique. This specific thesis work deals with the calculus of all the possible paths and reflections followed by a sound wave between the source and a given destination point, by using virtual transducers in an enclosed environment where reverberation exists. The aim of this work is to determine how to calculate that impulse response in a rectangular room in which the walls have uniform reflection coefficients and there isn’t any kind of furniture inside. Besides, this project specifies a simple way to make this calculus taking into account several sources and destinations.

  • 305.
    Oloumi, Daniel
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Ebadi, Siamak
    Kordzadeh, Atefeh
    Semnani, Abbas
    Mousavi, Pedram
    Gong, Xun
    Miniaturized Reflectarray Unit Cell Using Fractal-Shaped Patch-Slot Configuration2012In: IEEE Antennas and Wireless Propagation Letters, ISSN 1536-1225, E-ISSN 1548-5757, Vol. 11, p. 10-13Article in journal (Refereed)
    Abstract [en]

    This letter introduces a new class of miniaturized reflectarray unit cells with increased phase swing employing Minkowski fractal-shaped patch-slot elements. Square, 1st Minkowski, and 2nd Minkowski fractal patches are designed as a reflectarray unit cell. A slot with variable lengths of mm is used in the ground plane to perform the phase variation function. The resonant frequency corresponding to the maximum phase swing is reduced from 10.6 GHz for the square patch down to 8.8 and 8.3 GHz for the first-and second-order Minkowski fractal patches, respectively, which is equivalent to 17% and 22% size reduction. Unit cells with different patch type and slot length are fabricated, and close agreement is observed between the measured and simulated results. As it has been proven for conventional phased array antennas, this size reduction can lead to a decrease in mutual coupling in reflectarray antennas. Alternatively, it allows for smaller distance between reflectarray antenna elements, which renders a wider beam-scanning range.

  • 306.
    Oloumi, Daniel
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Moghadas, Hamid
    Mousavi, Pedram
    Dual-band orthogonally-polarized slotted-Lozenge reflective unit cell tuned by MEMS varactor2012Conference paper (Refereed)
    Abstract [en]

    A novel dual-band orthogonally-polarized MEMS reconfigurable slotted-Lozenge patch is introduced here. The dual-band performance of the lozenge patch (12 and 14 GHz) is due to different diagonal length. A cross slot is etched in the middle of patch to control the phase swing in each frequency and polarization. The effective slot length is controlled by MEMS capacitors with capacitance ratio of 1.5 resulting in phase swing of 345°. The reflect-array composed of this cell element would be able to radiate two independent beams at different frequencies and polarizations.

  • 307. Olsson, Sven
    Signal Processing as a Tool to Enhance Productivity in Industry: Measurement and Cancelation of Periodic Signals2006Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Preface This licentiate thesis summarizes my work within the field of Signal Processing during my time in Signal Control Sweden AB from 1987 to 2002. The work has been performed in close co-operation the customers like Volvo Personvagnar AB, Sveriges Provnings- och forskningsinstitut and others. Most of the material has been published at various conferences together with the Department of Signal Processing at Blekinge Institute of Technology. However, here I have had the opportunity to expand the text to give a more thorough description of the algorithms that have been designed. The thesis consists of five parts, which are: Part I An Order Analysis Method for Interior Noise Measurements in Cars Based on a Virtual Tachometer. II A Fast Multi-plane Shaft Balancing Method for 4-Wheel Drive Cars. III Custom Designed Filters through Time-Window Modification. IV Active Vibration Control of Cutting Operations. V Adaptive Vibration Control of a Large Dimension Carbon Fiber Boring Bar.

  • 308.
    Osman, Ammar
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Low-complexity OFDM transceiver design for UMTS-LTE2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Over the past two decades the mobile wireless communication systems has been growing fast and continuously. Therefore, the standardization bodies together with wireless researchers and mobile operators around the globe have been constantly working on new technical specifications in order to meet the demand for this rapid growth. The 3rd Generation Partnership Project (3GPP) one of the largest of such standardization bodies, works on developing the current third generation (3G) mobile telecommunication systems towards the future 4th generation. Research towards meeting the higher demands for higher data rates was the main reason for the birth of an evolution technology towards the 4th generation mobile systems. This evolution to the current 3rd generation UMTS systems was given the name E-UTRA/UTRAN Long Term Evolution (LTE) by the 3GPP. This thesis research has been carried out at the Telecommunications Research Center (ftw.) in Vienna. It was conducted in the framework of the C10 project “Wireless Evolution Beyond 3G”. One of the fields of research within this project is to have a special focus on the OFDM modulation schemes that are discussed under the new evolution technology (LTE) of the UMTS mobile networks. Therefore, this thesis focuses mainly in analyzing the new requirements, and evaluating them by designing a low-complexity UMTS-LTE OFDM based transceiver. This thesis aims mainly in studying the feasibility of this technology by means of simulation.

  • 309. Osman, Ammar
    et al.
    Mohammed, Abbas
    Performance Evaluation of a Low-complexity OFDM UMTS-LTE System2008Conference paper (Refereed)
    Abstract [en]

    Research towards meeting the higher demands for higher data rates was the main reason for the birth of an evolution technology towards the 4th generation mobile communication systems. This evolution to the current 3 rd generation UMTS systems was given the name E-UTRA/UTRAN Long Term Evolution (LTE) by the 3GPP. This paper analyzes the requirements for this evolution and evaluates the performance of the OFDM-LTE system under different propagation impairments (AWGN and multipath fading channels involving Pedestrian and Vehicular scenarios) in terms of bit and symbol error rates (BER and SER) for different modulation formats. © 2008 IEEE.

  • 310. Osman, Ammar
    et al.
    Mohammed, Abbas
    Yang, Zhe
    Multipath Wave Propagation Effects on the Performance of OFDM UMTS-LTE Communication System2008Conference paper (Refereed)
    Abstract [en]

    The proposed Long Term Evolution (LTE), for UMTS network, study by the Third Generation Partnership Project (3GPP) towards forth generation mobile telecommunication systems was one of the significant trends toward meeting the requirements and services of these future systems. This paper analyzes the requirements for this evolution and evaluates the performance of the Orthogonal Frequency Division Multiplexing (OFDM) UMTS-LTE system under different propagation impairments (AWGN, Pedestrian and Vehicular multipath fading channels of different speeds) in terms of bit and symbol error rates (BER and SER) for different modulation formats.

  • 311.
    Osunkunle, Isaac
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Shekarchi, Sayedali
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    A survey on methods for blind acoustic dereverberation2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Reverberation is a phenomenon in auditoriums such as concert halls and churches. Reverberation consists of a combination of multiple echoes, and its intensity and duration depend on factors such as the dimensions of the enclosure, materials used in construction and shape. Reverberation is desirable in music reproduction, however, it renders speech unintelligible. Thus there is a requirement to control reverberation of speech. This thesis work investigates the performances of different signal processing algorithms applied to suppress reverberation. Theoretical methods which have been verified with simulations are tested with real measurements. This gives a practical evaluation of the performance to be expected in the use of the algorithms.

  • 312.
    Ouyang, Dongfang
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Identification of Car Passengers with RFID for Automatic Crash Notification2009Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    Automatic Crash Notification is a system designed to be used in a crash situation. When a crash occurs, the intelligent system is activated and automatically sends select crash details to the appropriate Emergency Medical Service Center. These details can be the position of the vehicle and the likely severity of the damage. Using the information, the medical treatment resources demanded for the accident is assessed at Emergency Center. Accordingly, first-aid facilities are promptly and properly delivered to help the victims. Moreover, it would be a great advantage to include information about the passengers, such as the number of passengers, their age, sex and identity, in order to prepare the emergency services for their mission. The project focuses on implementation of Radio Frequency Identification technology (RFID) to improve the Crash Notification System of Autoliv Electronics together with First-Aid Profile (FAP). First-aid active RFID tag is pre-coded with a unique serial number (FAP-ID) that can be used to gain access to the First-Aid profile of that tagged person. Compatible reader detects the presence of First-aid tags and reports their FAP-IDs to Autoliv control unit, so that in crash situation, all passengers’ FAP-IDs will be messaged to Emergency Medical Service Center. During the project, the possibilities and constraints of using RFID technology for identifying passengers in vehicle is investigated, based on given hardware technological solution. Several tests are designed and carried out to investigate communication between the active RFID tag and the reader. Software program is also developed to build up the passenger identification system. According to experimental results, two possible implementations of the passenger identification system are proposed. Furthermore, the reliabilities of these two systems are tested against the situation when tag is buried in user’s pocket or bag.

  • 313.
    Parhizkari, Parvaneh
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Binaural Hearing-Human Ability of Sound Source Localization2009Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    Abstract The purpose of this project is to desig a systematical method in order to measure human directionality ability in horizontal plane with a single sound source. A completely virtual auditory model has been created in Matlab. The project consists of modeling binaural cues, designing digital filters, designing a test workbench, measuring listener's directionality and analyzing the data. The head related transfer function (HRTF) is computed by calculating the two most important binaural cues, interaural level difference (ILD) and interaural time difference (ITD). The platform is made in Matlab and all results have been shown by plots produced from Matlab code. The directionality test has been done with real human subjects and the results have been analyzed and presented.

  • 314.
    Peretz, Avi
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Gumusoglu, Cagdas
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    G.722 WIDEBAND SPEECH CODEC IMPLEMENTATION ON BF-533 DSP2004Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    The technological developments in digital communication systems increase the usable bandwidth of sound signals which results in increased intelligibility and naturalness of the signal. The emerging digital communication systems enable the use of wideband speech codec in a wide area of applications. Recognizing the need of high quality wide band speech codec, several standardization activities have been recently conducted. G.722 is one of the first wideband speech codec standards implemented in the telecommunication systems. In this thesis, after examining the G.722 wideband speech codec theoretically, a real time implementation of G.722 wideband speech codec has been developed on Blackfin BF533 DSP development kit from Analog Devices. Also, G.722 Codec is optimized by using assembler for BF533. As a result to the thesis, very good speech performance is obtained in real time implementation of G.722 and optimization increases the eciency of Blackfin BF533 DSP drastically.

  • 315.
    Persson, Thom
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Building of a Stereo Camera System2009Independent thesis Basic level (degree of Bachelor)Student thesis
    Abstract [en]

    This project consists of a prototype of a stereo camera rig where you can mount two DSLR cameras, and a multithreaded software application, written in C++, that can move the cameras, change camera settings and take pictures. The resulting 3D-images can be viewed with a 2-view autostereoscopic display. Camera position is controlled by a step engine which is controlled by a PIC microcontroller. All communication with the PIC and the computer is made over USB. The camera shutters are synchronized so it is possible to take pictures of moving objects at a distance of 2.5 m or more. The results shows that there are several things to do before the prototype can be considered a product ready for the market, most of all the camera callback functionality.

  • 316.
    Petermann, Patrik
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Förbättrad resistansmätning för grova aluminiumledare2005Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Målet med examensarbetet har varit att utvärdera och undersöka den stationära mätbryggan. Mätbryggan använder en 4-punktsmätning för bestämning av resistansen som används i ledartillverkningen, och används för grövre aluminiumledare. En ledare ska uppfylla vissa krav för att gå vidare i processen, ett av dessa är resistansen i ohm/km. Vilket ställer krav på att mätbryggans mätvärde är så sant och noggrant som möjligt. Den stationära mätbryggan har undersökts genom olika mätningar för att se hur stabilt systemet är och hur ett så bra mätvärde ska tillhandahållas. Mätningarna utfördes på olika dimensioner och typer av aluminiumledare, i varje mätning utfördes ett antal olika test för aluminiumledaren. Resultat av mätningarna visar att systemet inte är stabilt för aluminiumledare med tvärsnittsarea över ~1000 mm2. Detta beror på att strömmen i mätbryggan inte fördelar sig jämnt i hela ledaren tvärsnitt. Men genom att använda sig av kontaktpressning har strömfördelning förbättrats och medför ett stabilare mätvärde. Men för aluminiumledare med svällpulver och svällband kommer det att behövas en annan metod än kontaktpressning för att bestämma resistansen i ohm/km.

  • 317. Pettersson, Mats
    Detection of Moving Targets in Wideband SAR2004In: IEEE Transactions on Aerospace and Electronic Systems, ISSN 0018-9251, E-ISSN 1557-9603, Vol. 40, no 3, p. 780-796Article in journal (Refereed)
    Abstract [en]

    A likelihood ratio is proposed for moving target detection in a wideband (WB) synthetic aperture radar (SAR) system. WB is defined here as any systems having a large fractional bandwidth, i.e., an ultra wide frequency band combined with a wide antenna beam. The developed method combines time-domain fast backprojection SAR processing methods with moving target detection using space-time processing. The proposed method reduces computational load when sets of relative speeds can be tested using the same clutter-suppressed subaperture beams. The proposed method is tested on narrowband radar data.

  • 318. Pettersson, Mats
    Four-Dimensional Discretization for Detection of Moving Objects in Wide Band SAR2007Conference paper (Refereed)
    Abstract [en]

    This paper addresses four-dimensional Ground Moving Target Indication (GMTI) using a multi-channel Wide Band (WB) Synthetic Aperture Radar (SAR) system. With no acceleration attached to the target the motion of the target is related to two ground and two speed coordinates. However, four other dimensions are used in WB SAR GMTI processing during the detection phase: azimuth, range, bearing, and the relative speed between the object and the SAR platform. In the detection phase, blind hypotheses are used, and the discretization steps between the hypotheses are a trade-off between the number of hypotheses tested and detectability. As the integration angle increases, the step size in the image dimensions and in the relative speed has to be reduced. In this paper we determine the discretization step in all four dimensions for moving target detection, and relate it to radar system parameters. The discretization is derived from the moving target impulse response, assuming independency between the dimensions. In the paper the number of hypotheses per square meter is given for an airborne low frequency and a microwave GMTI WB SAR system.

  • 319. Pettersson, Mats
    Lower Bounds of Moving Target Estimation in Low Frequency SAR2008Conference paper (Refereed)
    Abstract [en]

    Lower bound of moving target estimation, in a multi-antenna channel Ultra Wide Band (UWB) Synthetic Aperture Radar (SAR) system is studied in this work. The main focus is to derive lower bound for systems having a very long integration time associated with position and velocity estimation, which is often connected to low frequency systems. Assuming a flat Earth model we consider the lower bound of the speed components in range and in the cross range direction. The estimation in cross range direction is associated with a considerable integration angle, and therefore the lower bound is determined in connection to a SAR process. It is found that the relative speed influences the covariance matrix of the clutter and therefore also the speed estimate. The influence increases the estimation performance of the relative speed and therefore also the performance of the cross range speed estimation.

  • 320. Pettersson, Mats
    Optimum relative speed discretisation for detection of moving objects in wide band SAR2007In: IET Radar, Sonar and Navigation, ISSN 1751-8784 , Vol. 1, no 3, p. 213-220Article in journal (Refereed)
    Abstract [en]

    Here, Ground moving target indication (GMTI) using synthetic aperture radar (SAR) is considered. SAR GMTI requires that relative speed between the target and the SAR platform is included in the detection algorithm. A separation between the true relative speed and the relative speed used in the SAR process will cause unfocused targets, and decrease detectability. Blind hypotheses of relative speeds are used in the detection phase of moving targets in SAR. The step size between the hypotheses (or discretisation step) in relative speed involves a trade off between the number of hypotheses to test and detectability. A large number of tests will increase detectability but will also increase computation load and vice versa. The relevance of relative speed increases as the azimuth integration time gets larger. Long integration time is associated with low signature moving target detection in strong clutter environments, or for SAR GMTI at low frequencies. The optimum discretisation of normalised relative speed for moving target detection has been determined. The optimum discretisation is derived from the moving target impulse response. Use of optimum discretisation reduces the computation burden in SAR GMTI and secures the detectability.

  • 321. Pettersson, Mats
    Relative Speed Step Size in SAR processing for Moving Target Detection2006Conference paper (Refereed)
    Abstract [en]

    Ground Moving Target Indication (GMTI) using Synthetic Aperture Radar (SAR) is studied in this paper. For systems using long integration time Relative Speed between the target and the SAR platform has to be included in the detection algorithm. A separation between the true Relative Speed and the Relative Speed used in the SAR process will cause unfocused targets. Unfocused targets decrease the detectability. In the detection phase of SAR moving targets blind hypothesis on relative are used. The step size between the hypotheses or the quantization step in Relative Speed is a trade off between the number of hypotheses to test and detectability. A large number of tests will increase detectability but will also increase computation load and vice versa. The importance of relative speed increases as the azimuth integration time gets larger. Long integration time is associated with low signature moving targets detection in strong clutter environment and especially for SAR GMTI at low frequencies. In this paper we determine the optimum quantization of relative speed for moving target detection. The optimum quantization is derived from the moving target impulse response. By using the optimum quantization the computation burden in SAR GMTI is reduced and the detectability secured.

  • 322. Pettersson, Mats
    et al.
    Vu, Viet Thuy
    Sjögren, Thomas
    Gustavsson, Anders
    Multi-Dimensional Hypotheses Test of Movement Detection in Wide Band Radar Systems Associated with Long Integration Time2008Conference paper (Refereed)
    Abstract [en]

    This paper addresses multi-dimensional Ground Moving Target Indication (GMTI) using a multi-channel Wide Band (WB) Synthetic Aperture Radar (SAR) system. For limited time intervals the target acceleration is so small that target motion can be related to two ground and two speed coordinates. However, four other dimensions are used in WB SAR GMTI processing during the detection phase: azimuth, range, bearing, and the relative speed between the object and the SAR platform. In the detection phase, blind hypotheses are used, and the discretization steps between the hypotheses are a trade-off between the number of hypotheses tested and detectability. As the integration angle increases, the bandwidth increases and the therefore the number of tests increases. In this paper we discuss the discretization step in all four dimensions for moving target detection, and analyze the step size in particular in the most critical domain, the relative speed. The analysis is made on CARABAS II data.

  • 323. Pettersson, Mats
    et al.
    Zetterberg, Viktoria
    Claesson, Ingvar
    Detection and imaging of moving targets in wide band SAS using fast time backprojection combined with space time processing2005Conference paper (Refereed)
    Abstract [en]

    This paper present a method to combine SAS (Synthetic Aperture Sonar) imaging of stationary targets with moving target detection and imaging. The proposed method uses a likelihood ratio for moving target detection in a wide band (WB) SAS system. For this paper, WB is defined as any systems having a large fractional bandwidth, i.e. an ultra wide frequency band combined with a wide antenna beam. The developed method combines time domain fast backprojection SAS processing methods with moving target detection using space-time processing. In the paper defocusing and detection of moving targets are investigated. Both the trajectory and the location of the moving target is given by mathematical expressions.

  • 324.
    Prasad, Kamtala Venkat Ramana
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Modeling of Broadband Shortwave Antennas With Numerical Electromagnetics Code2009Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    Antenna designers always search for ways to improve existing designs or introduce novel designs in order to achieve desirable antenna characteristics. The linear wire dipole antennas are very important in communication systems at all frequency bands. These antennas are used by typically military, navigation and surveillance purposes. The broadband antenna which was built in the year of 1950 and was working efficiently for over half a century, some problems such as inefficient radiation, imperfect feeding point impedance and high voltage standing wave ratio (VSWR) were discovered when sending information via this broadband antenna in the operating frequency range of 9-11 MHz. In order to meet the proper specifications antenna is modeled before implementing physically. The Numerical Electromagnetics Code (NEC-4) is used to simulate the antenna models and results are analyzed by using MATLAB. The results of this study indicate that the antenna models achieve more than 70% of radiation efficiency including practical considerations such as ground parameters, material of the antenna and height of the antenna, impedance matching and low VSWR.

  • 325.
    Radhakrishnan, Krishnakumar
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Implementation of a Soft output sphere decoder by rapid prototyping methhodology2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    To meet the demand for faster wireless access there is a need to change from the traditional Single-Input-Single Output (SISO) antenna systems.Multiple-input-multiple-output (MIMO) uses multiple antennas in both transmitter and receiver ends to achieve high spectral efficiency. However the implementation of MIMO systems comes at the cost of increased complexity in the receiver design. The detection of vector of symbols drawn from a finite alphabet when transmitted over a MIMO system poses a challenging implementation task. The aim of this thesis is to use sphere detection algorithm as an efficient means to detect the spatially multiplexed symbols. This thesis discusses the evaluation of the performance and implementation aspects of a Soft output sphere decoder as an efficient detection means for MIMO systems.

  • 326.
    Radhakrishnan, Krishnakumar
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Implementation of Soft-output sphere decoder by rapid prototyping methodology2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    To meet the demand for faster wireless access there is a need to change from the traditional Single-Input-Single Output (SISO) antenna systems.Multiple-input-multiple-output (MIMO) uses multiple antennas in both transmitter and receiver ends to achieve high spectral efficiency. However the implementation of MIMO systems comes at the cost of increased complexity in the receiver design. The detection of vector of symbols drawn from a finite alphabet when transmitted over a MIMO system poses a challenging implementation task. The aim of this thesis is to use sphere detection algorithm as an efficient means to detect the spatially multiplexed symbols. This thesis discusses the evaluation of the performance and implementation aspects of a Soft output sphere decoder as an efficient detection means for MIMO systems.

  • 327.
    Rakus-Andersson, Elisabeth
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Mathematics and Science.
    Zettervall, Hang
    Blekinge Institute of Technology, School of Engineering, Department of Mathematics and Science.
    Erman, Maria
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Prioritisation of weighted strategies in multiplayer games with fuzzy entries of the payoff matrix2010In: International Journal of General Systems, ISSN 0308-1079, E-ISSN 1563-5104, Vol. 39, no 3, p. 291-304Article in journal (Refereed)
    Abstract [en]

    We explore the classical model of a two-player game to select the best strategies, where action is expected to maintain the values of a certain variable on the neutral level. By inserting fuzzy sets as payoff values in the game matrix, we facilitate the procedure of formulations of payoff expectations by players. Instead of making inconvenient decisions about the choice of accurate numerical entries of the matrix, the players are able to use words, which should simplify communication between them when designing the preliminaries of the game. The players also have the possibility of making a ranking of their favourite strategies. At the next stage of the play, we involve group decision-making in order to aggregate results coming from several paired games, when more than two players contradict each other.

  • 328.
    RAO, THUMATI VENKATA
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    DATA PROCESSING DESIGN OF WIDEBAND CODE DIVISION MULTIPLE ACCESS (WCDMA)2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Wideband Direct-Sequence Code Division Multiple Access (W-CDMA) is the emerging protocol for the next generation (3G) wireless communication systems. W-CDMA has been designed to add features such as multimedia capabilities, high data rates and multi-rate services to the existing wireless communication framework. Several standards for third generation systems have been proposed and developed by different industrial committees in countries such as the U.S, Europe and Japan. All these standards have accepted CDMA in one form or another as the multiple access method for wireless communications requirements. In this work, we study the implementation issues involved for one of the proposed Rate Compatible Punctured Convolution Coding (RCPC) to increase information rate in WCDMA, the detection algorithms for base-stations. It was found that these proposed algorithms for detection have different processing and precision requirements. In the rate compatible punctured convolution from each coded symbol some of the bits are punctured in order to achieve the higher information rate. At the same time decoding becomes quite easy even if some of the bits from the coded symbol are not available. We have written these algorithms for a single user at ideal conditions based on theory but not practically.

  • 329. Rohani, Behrooz
    et al.
    Rohani, Bijan
    Caldera, Manora
    Zepernick, Hans-Jürgen
    Benefits of Perceptual Speech Quality Metrics in Modern Cellular Systems2006In: Electronics Letters, ISSN 0013-5194, E-ISSN 1350-911X, Vol. 42, no 21, p. 1250-1251Article in journal (Refereed)
    Abstract [en]

    Advanced algorithms have become available in recent years that can reliably measure the speech quality as “perceived” by humans. The benefits of applying the perceptual quality measures obtained using these algorithms in the Outer Loop Power Control (OLPC) of Third Generation Universal Mobile Telecommunication System (3G UMTS) are studied in this letter. It is shown that 20% capacity improvement compared to the use of conventional measures can be achieved while an adequate and uniform speech quality is maintained.

  • 330. Rohani, Behrooz
    et al.
    Rohani, Bijan
    Caldera, Manora
    Zepernick, Hans-Jürgen
    On In-service Perceptual Speech Quality Monitoring in Cellular Radio Systems2009Conference paper (Refereed)
    Abstract [en]

    A method for in-service monitoring of the end-user perceptual speech quality in cellular radio systems is proposed. This method incorporates the perceptual evaluation of speech quality (PESQ) algorithm to monitor the quality experienced by the end-user. Here, the monitoring is carried out at the transmitting side. In this case, the speech signal received by the end-user is estimated at the transmitter in accordance with a feedback signal. The performance of the proposed scheme has been investigated through extensive simulations for the Universal Mobile Telecommunication System (UMTS) using different speech coding rates and channel conditions. The results indicate that the proposed scheme can predict end-user quality with a root-mean-squared error (RMSE) of at most 0.15 using the mean opinion score (MOS) rating scheme. Such accuracy can be beneficial in applications such as radio resource management for satisfying the desired level of quality of service.

  • 331. Rohani, Behrooz
    et al.
    Rohani, Bijan
    Zepernick, Hans-Jürgen
    Feedback Method for Real-time Perceptual Quality Estimation2004In: Electronics Letters, ISSN 0013-5194, E-ISSN 1350-911X, Vol. 40, no 14, p. 913-914Article in journal (Refereed)
    Abstract [en]

    In this paper, a feedback scheme for real-time estimation of perceptual speech quality is proposed. The feedback information indicates the quality of each previously received speech frame as “good” or “bad”. This information is used in conjunction with the PESQ algorithm to estimate the received speech quality accurately.

  • 332. Rohani, Bijan
    et al.
    Zepernick, Hans-Jürgen
    Erlang Capacity of UMTS Using Perceptual-based Power Control: An Analytical Framework2009Conference paper (Refereed)
    Abstract [en]

    An analytical framework for calculation of Erlang capacity per cell of UMTS with perceptual-based power control is derived. A comparison of the capacity of the system using the proposed power control algorithm with its conventional counterpart is performed. This is supported by numerical results on Erlang capacity over a variety of system parameters for both considered power control approaches. It is shown that the use of perceptual speech quality metrics in power control results in a capacity gain of at least 10% over that of conventional UMTS.

  • 333.
    Rosen, Anders
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Nilsson, Kristian
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Control System for a Remote Electronics Laboratory2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    The work done in this thesis has to do with the equipment server which has been upgraded with a new circuit wiring robot and a new voltage source and therefore needed new software. The main change were that the old circuit wiring robot, or matrix, were controlled via a IO card that were cumbersome to use and rather expensive, this was changed to a USB interface which is more generic and relatively easy to use.

  • 334. Rossholm, Andreas
    On the Enhancement of Audio and Video in Mobile Equipment2006Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Use of mobile equipment has increased exponentially over the last decade. As use becomes more widespread so too does the demand for new functionalities. The limited memory and computational power of many mobile devices has proven to be a challenge resulting in many innovative solutions and a number of new standards. Despite this, there is often a requirement for additional enhancement to improve quality. The focus of this thesis work has been to perform enhancement within two different areas; audio or speech encoding and video encoding/decoding. The audio enhancement section of this thesis addresses the well known problem in the GSM system with an interfering signal generated by the switching nature of TDMA cellular telephony. Two different solutions are given to suppress such interference internally in the mobile handset. The first method involves the use of subtractive noise cancellation employing correlators, the second uses a structure of IIR noth filters. Both solutions use control algorithms based on the state of the communication between the mobile handset and the base station. The video section of this thesis presents two post-filters and one pre-filter. The two post-filters are designed to improve visual quality of highly compressed video streams from standard, block-based video codecs by combating both blocking and ringing artifacts. The second post-filter also performs sharpening. The pre-filter is designed to increase the coding efficiency of a standard block based video codec. By introducing a pre-processing algorithm before the encoder, the amount of camera disturbance and the complexity of the sequence can be decreased, thereby increasing coding efficiency.

  • 335. Rossholm, Andreas
    et al.
    Andersson, Kenneth
    Adaptive De-Blocking De-Ringing Post Filter2005Conference paper (Refereed)
    Abstract [en]

    In this paper an adaptive filter for reducing blocking and ringing artifacts is presented. The solution is designed with consideration of Mobile Equipment with limited computational power and memory. Also, the solution is computationally scalable if there is limited CPU resources in different user cases.

  • 336. Rossholm, Andreas
    et al.
    Lövström, Benny
    A New Low Complex Reference Free Video Quality Predictor2008Conference paper (Refereed)
    Abstract [en]

    In many applications and environments for mobile communication there is a need for reference free perceptual quality measurements. In this paper a method for prediction of a number of quality metrics is proposed, where the input to the prediction is readily available parameters at the receiver side of a communications channel. Since the parameters are extracted from the coded video bit stream the model can be used in user scenarios where it is normally difficult to estimate the quality due to the reference not being available, as in streaming video and mobile TV applications. The predictor turns out to give good results for both the PSNR and the PEVQ metrics.

  • 337. Rossholm, Andreas
    et al.
    Lövström, Benny
    A New Video Quality Predictor Based on Decoder Parameter Extraction2008Conference paper (Refereed)
    Abstract [en]

    In the mobile communication area there is a demand for reference free perceptual quality measurements in video applications. In addition low complexity measurements are required. This paper proposes a method for prediction of a number of well known quality metrics, where the inputs to the predictors are readily available parameters at the decoder side of the communication channel. After an investigation of the dependencies between these parameters and between each parameter and the quality metrics, a set of parameters is chosen for the predictor. This predictor shows good results, especially for the PSNR

  • 338. Rossholm, Andreas
    et al.
    Lövström, Benny
    Andersson, Kenneth
    Low-Complex Adaptive Post Filter for Enhancement of Coded Video2007Conference paper (Refereed)
    Abstract [en]

    In this paper an adaptive filter that removes de-blocking and de-ringing artifacts and also enhances the sharpness of decoded video, which may be caused by zeroing high-frequency DCT coefficients, is presented. The solution is designed with consideration of Mobile Equipment with limited computational power and memory. Also, the solution is computationally scalable to be able to handle limited computational resources in different user cases. In the paper it is shown that the adaptive filter always keeps or increases the image quality, compared to the original decoded sequences, and that the amount of sharpening decreases with an decrease of bit-rate to limit amplification of coding artifacts or noise.

  • 339. Scheuer, Johan
    et al.
    Håkansson, Lars
    Claesson, Ingvar
    Modal Analysis of a Boring Bar using Different Clamping Conditions2004Conference paper (Refereed)
    Abstract [en]

    The internal turning operation is a common metal working process that is usually associated with vibration problems. Boring bars are often long and slender, making them prone to vibration when they are excited by the cutting forces introduced by the material deformation process in boring. The vibration causes degraded surface finish of the machined work piece, decreased tool life of inserts and boring bars, noise and other unwanted effects. Therefore, time-consuming planning and preparation have to be made in order to minimize vibration, affecting the cost of the machining process in the negative direction. It is possible to stabilize boring bars using active vibration control. To be able to do so, the dynamic properties of the bars have to be known in detail. Furthermore, with this knowledge, the results of using prediction tools for relating cutting parameters to cutting results may be significantly improved thanks to more relevant input data into the prediction tools. The effect of different clamping conditions on the dynamic properties of a boring bar's fundamental bending modes has been studied. Modal analysis was performed on a 40 mm-diameter boring bar with 100 mm clamping length and 200 mm overhang, making the L/D ratio 5. The clamping conditions have been varied by use of two different clamping devices. In each clamping device, several different clamping conditions were achieved by using different torque on the clamping screws. Results show that both the eigenfrequencies and the directions of the fundamental bending modes vary considerably due to the clamping conditions used.

  • 340. Schüldt, Christian
    Low-Complexity Adaptive Filtering for Acoustic Echo Cancellation in Audio Conferencing Systems2009Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    With the globalization of the world’s economy, the demand for effortless, quick and efficient communication is increasing. Modern audio conferencing allows people at different locations to have a conversation as if they were sitting in the same room, without having to travel. This obviously saves time and money, and also lessens the environmental strain caused by travel. Most audio conferencing systems and hands-free systems in particular, suffer from electric and/or acoustic echoes. Electric echoes typically originate from 2-4 wire conversion in hybrid circuits in the telephone network, while acoustic echoes arise due to acoustic coupling between loudspeaker and microphone. In digital audio communication equipment, the echoes are usually removed through digital signal processing methods such as adaptive filtering. Since audio conferencing systems are consumer electronic products, the manufacturing cost is a key issue. In order to accomplish low manufacturing costs, the choice of a low cost digital signal processor (DSP) to perform the signal processing tasks is central. Further, due to the limited resources of low cost DSPs, there is an intrinsic demand for low complexity signal processing algorithms. This thesis presents low complexity algorithms for adaptive filtering in acoustic echo cancellation applications. Both the actual update of the adaptive filter and the update control to prevent divergence and so called howling, are considered. Computer simulations, as well as real time implementations in actual acoustic systems are used to verify the performance of the proposed algorithms.

  • 341. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    A Combined Implementation of Echo Suppression, Noise Reduction and Comfort Noise in a Speaker Phone Application2007Conference paper (Refereed)
    Abstract [en]

    Echo suppression, noise reduction and comfort noise are desirable features in loudspeaker phone products. This paper proposes a set of algorithms for a combined, subband based, implementation of these three processing blocks. The proposed algorithms are verified by evaluation of a fix-point real-time implementation.

  • 342.
    Schüldt, Christian
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    Lindström, Fredric
    Claesson, Ingvar
    Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    A Delay-Based Double-Talk Detector2012In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 20, no 6, p. 1725-1733Article in journal (Refereed)
    Abstract [en]

    When an adaptive filter is used for echo cancellation, it is essential to prevent the filter from diverging in situations when the echo signal is contaminated with near-end disturbance, i.e. during double-talk. This paper presents an extension of a previously proposed double-talk detector for improved performance. It is shown that the computational complexity of the proposed detector is lower than that of the well-used normalized cross correlation (NCC) double-talk detector, at the cost of performance. Further, it is shown that there can be a significant performance difference, in terms of detecting double-talk, between having a fixed echo cancellation filter, which is a common strategy in objective evaluation techniques, and an adaptive filter, which is more close to realistic conditions.

  • 343. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    A Distortion Reducing Subband Limiter Implementation for Conference Phones2008Conference paper (Refereed)
    Abstract [en]

    Distortion, most often caused by the loudspeaker, is a practical problem in acoustic echo cancellation-based conference phones. Typically, this distortion varies significantly with the frequency. This paper presents an implementation of a distortion reducing subband limiter to be used in conjunction with an acoustic echo canceller in a commercial conference phone, yielding improved cancellation performance. Verification of the performance was performed through evaluation of a fix-point real-time implementation.

  • 344. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    A Low-Complexity Delayless Selective Subband Adaptive Filtering Algorithm2008In: IEEE Transactions on Signal Processing, ISSN 1053-587X, E-ISSN 1941-0476, Vol. 56, no 12, p. 5840-5850Article in journal (Refereed)
    Abstract [en]

    Adaptive filters of significant order, requiring high computational complexity, are necessary in many applications such as acoustic echo cancellation and wideband active noise control. Successful approaches to lessen the computational complexity of such filters are subband methods, and partial updating schemes where only a part of the filter is updated at each instant. To avoid the time delay introduced by the subband-splitting, delayless structures which reconstructs a fullband filter, producing delayless output, from the adaptive subband filters have been proposed. This paper proposes a delayless subband adaptive filter partial updating scheme, where the general idea is to only update the most misadjusted subband filter(s). Analysis in terms of mean square deviation is presented and shows that the fullband filter convergence speed is significantly increased, even for flat spectrum signals, as compared to traditional periodic subband filter update with the same computational complexity. Echo cancellation simulations with an artificial system to verify the analysis, using both flat spectrum signals and speech, is also presented, as well as offline calculations using signals from a real system.

  • 345. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    Evaluation of an Improved Deviation Measure for Two-Path Echo Cancellation2010Conference paper (Refereed)
    Abstract [en]

    The two-path algorithm is a well-known approach for overcoming the dead-lock problem in echo cancellation systems. Typically, a fixed foreground filter is producing the echo cancelled output while a continuously updating background filter adapts to the echo-path. When the background filter is considered to perform better than the foreground filter, the coefficients of the background filter are copied into the foreground filter. To determine which filter is better adjusted to the true echo-path, a filter deviation measure can be used. Recently, a method which introduces a delay in the calculation of the filter deviation measure, yielding a more reliable estimate has been proposed. However, a thorough evaluation of the effect of different delay settings has not yet been performed. Thus, in this paper a number of simulations with different delay parameter settings are carried out to show how this parameter affects the overall performance of the filter deviation measure.

  • 346. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    Low-Complexity Adaptive Filtering Implementation for Acoustic Echo Cancellation2006Conference paper (Refereed)
    Abstract [en]

    Acoustic echo cancellation is generally achieved with adaptive FIR filters. Due to the often large dimensionality of the adaptive filters, required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a block based selective updating method which reduces the complexity with nearly a half in practical situations, while showing superior convergence speed performance as compared to conventional partial update complexity reduction schemes.

  • 347. Schüldt, Christian
    et al.
    Lindström, Fredric
    Claesson, Ingvar
    Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    Robust low-complexity transfer logic for two-path echo cancellation2012Conference paper (Refereed)
    Abstract [en]

    A well used approach for echo cancellation is the two-path method, where two adaptive filters in parallel are utilized. Typically, one filter is continuously updated, and when this filter is considered better adjusted to the echo-path than the other filter, the coefficients of the better adjusted filter is transferred to the other filter. When this transfer should occur is controlled by the transfer logic. This paper proposes transfer logic that is both more robust and more simple to tune, owing to fewer parameters, than the conventional approach. Extensive simulations show the advantages of the proposed method.

  • 348. Schüldt, Christian
    et al.
    Lindström, Fredric
    Li, HB
    Adaptive filter length selection for acoustic echo cancellation2009In: Signal Processing, ISSN 0165-1684, E-ISSN 1872-7557, Vol. 89, no 6, p. 1185-1194Article in journal (Refereed)
    Abstract [en]

    The number of coefficients in an adaptive finite impulse response filter-based acoustic echo cancellation setup is an important parameter, affecting the overall performance of the echo cancellation. Too few coefficients give undermodelling and too many cause slow convergence and an additional echo due to the mismatch of the extra coefficients. This paper proposes a method to adaptively determine the filter length, based on estimation of the mean square deviation. The method is primarily intended for identifying long non-sparse systems, such as a typical impulse response from an acoustic setup. Simulations with band limited flat spectrum signals are used for verification, showing the behavior and benefits of the proposed algorithm. Furthermore, off-line calculation using recorded speech signals show the behavior in real situations and comparison with another state-of-the-art variable filter length algorithm shows the advantages of the proposed method. (c) 2009 Elsevier B.V. All rights reserved.

  • 349. Seun, Ajayi Taiwo
    et al.
    Mohammed, Abbas
    Yang, Zhe
    Solomonsson, Maria
    Channel Modelling and Characterization for Mobile Satellite Communication Systems2009Conference paper (Refereed)
    Abstract [en]

    Reliable characterization of the propagation environment and channel modelling of mobile satellite communication systems is necessary in order to provide better quality of service and efficient design of these systems. In this paper, the channel impairments affecting the performance and an overview of the satellite channel models are presented. The statistical distributions of the received signal that can be used to characterize the dynamic nature of these propagation channels are also presented. The main modelling parameters are investigated and simulation results show that the bit error rate performance (BER) is predominantly affected by the shadowing factor.

  • 350.
    Shabbir, Noman
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Kasif, Hasnain
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Radio Resource Management in WiMAX2009Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    Today, different types of cellular networks are actively working on the radio links. For instance, the Global System for Mobile Communication (GSM) is being used in nearly two hundred countries and currently it has around two and half billion users all over the world. Universal Mobile Telecommunication System (UMTS) is currently deployed in many countries and it is providing increased data rates, coverage and mobility as compared to GSM. Wireless Local Area Networks (WLAN) are very famous when we have a small area and none real time services. Worldwide Interoperability for Microwave Access (WiMAX) is a new technology and it is in deployment phase. In all these cellular technologies, we have very limited recourses and we have to make best use of them by proper management. Radio Resource Management (RRM) is a control mechanism for the overall system which is being used to manage radio resources in the air interface inside a cellular network. The main objective is to utilize the available spectral resources as efficiently as possible. Our aim is to use them in the best possible way to maximize the performance and spectral efficiency in such a way that we have maximum number of users in our network and Quality of Service (QoS) is up to the mark. In a cellular communication system, a service area or a geographical region is divided into a number of cells and each cell is served by an infrastructure element called the base station which works through a radio interface. The frequency license fees, real estate, distribution network and maintenance are the issues which dominates the cost for deploying a cellular network. Management of radio related resources is a critical design component in cellular communications. In RRM, we control parameters like Radio Frequency (RF) planning, link budgeting, modulation schemes, channel access schemes etc. RF planning includes cell planning, coverage of the network and capacity of the network. Our main focus in this thesis will be on cell planning and link budgeting and we will discuss them in context of a WiMAX network.

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