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  • 201. Khan, Muhammad Gufran
    et al.
    Sällberg, Benny
    Nordberg, Jörgen
    Claesson, Ingvar
    Robust Weighted Non-Coherent Receiver for Impulse Radio UWB PPM Signals2011In: IEEE Communications Letters, ISSN 1089-7798, E-ISSN 1558-2558, Vol. 15, no 06, p. 614-616Article in journal (Refereed)
    Abstract [en]

    This letter proposes an energy detection based robust weight estimation scheme for pulse-position modulated (PPM) impulse radio ultra-wideband (IR-UWB) signals using weighted non-coherent receiver (WNCR). Conventional data-aided WNCR (DA-WNCR) scheme estimates the weighting coefficients, or channel state information (CSI), using a large number of training symbols over time-varying channels. In contrast, the proposed Robust WNCR (R-WNCR) scheme is non-data-aided (NDA), adaptive and robust to channel variations. The proposed R-WNCR estimates the weighting coefficients adaptively based on the received stochastic data, and the weight estimation process is refined using a decision directed approach.

  • 202. Khan, Nazmul
    et al.
    Alam, Mohammad
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Noise reduction algorithm for LS channel estimation in OFDM system2012Conference paper (Refereed)
    Abstract [en]

    Wireless communication system incorporating coherent OFDM requires the estimation and tracking of the channel Impulse Response (CIR) for accurate demodulation of data at the receiver. In pilot-symbol-aided OFDM system, Minimum Mean Square Error (MMSE) estimator performs better than Least Square (LS) estimator; however, computational complexity associated with the MMSE method is relatively higher than the LS. Although, the LS estimator has lower complexity and requires minimum knowledge the channel state information, the estimator suffers from inherent additive Gaussian noise and Inter Carrier Interference (ICI). Following that, in this study an efficient and improved channel estimation technique is proposed based on the LS algorithm. Simulation results show that the proposed method performs considerably better than the conventional LS method for a range of Signal to Noise Ratios (SNRs). In addition, the performance of the proposed method is found to be almost equal, if compared with the MMSE estimator. Despite the proposed method experiences relatively higher computational complexity than the LS, the complexity is yet to be achieved about 40 % lower than the MMSE.

  • 203.
    Khan, Shabbar Ali
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    A study of closed loop space time block coding techniques for MIMO-OFDM2006Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    In this dissertation, the STBC-OFDM technique is therefore investigated in order to improve the bit error rate performance of such wireless communication systems speci¯cally those using multiple transmitter and receiver antennas.An investigation is undertaken of STBC with two transmit and one receive antennas as proposed by Alamouti together with quasi-orthogonal STBC with four transmit and one receive antennas. The simulated performance shows that both full diversity and full code rate is achieved only in the case of two transmit and one receive antenna, whereas the diversity is decreased in the case of four transmit and one receive antennas due to the presence of coupling term. This dissertation also presents a method of minimizing such coupling terms by exploiting a phase feedback method as proposed by Cenk Toker, which forces the coupling terms to zero. Both one phase feedback and two phase feedback methods are used. The simulated performance shows that the practical performance gain between three bit feedback and the two bit feedback is only 0.43dB, so to use two bit feedback appears appropriate. Only a ¯nite number of bits is allowed for the feedback, to satisfy the quasi-static assumption on the channel. Finally, closed loop space time block code-OFDM is also investigated, since OFDM is well known to be suited for high data rate applications in fading channels, due to its high spectral efficiency. Another advantage of an OFDM system lies in its ability to transfer a frequency-selective fading channel into multiple °at fading channels on different sub-carriers. The simulation results obtained clearly show that the performance gain between three bit feedback and the two bit feedback is 0.45 dB, therefore a practical two-bit feedback scheme is feasible.

  • 204.
    Khan, Sohaib
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Cuong, Nguyen Kim
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    HSDPA System Simulation2005Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    This thesis provides a background of the high speed downlink packet access (HSDPA) concept; a new feature which has been introduced in the Release 5 specifications of the 3GPP WCDMA/UTRA-FDD standards. In order to emphasize the theoretical analysis, a simulation of a HSDPA system shall also be performed.

  • 205.
    Khan, Zaheer
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Performance Analysis of Spreading Sequences in Radio Channels2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    In this thesis, we have studied direct sequence spread spectrum based CDMA communication system. We have analyzed CDMA based transmitter and receiver structures operating in Rayleigh fading and additive white Gaussian noise (AWGN) channel characteristics, in which data is transmitted and received. We next focused on spreading sequences used in such systems and analyzed the system performance in the presence of various sequences. We have studied the sequence properties, analyzed their bit error rate (BER) and compared traditional Walsh Hadamard sequences with uni¯ed complex Hadamard transform (UCHT) sequences, modi¯ed UCHT sequences and complex Oppermann sequences. We concluded that complex Oppermann se- quences and some UCHT sequences outperform Walsh Hadamard sequences in asynchronous CDMA systems because of their polyphase characteristics and better correlation properties. Next, we considered the problem of optimization among the sequence sets and showed that optimized subsets perform significantly better as compared to random selection of sequences within a sequence set.

  • 206.
    Khatir, Kamal
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Coded Cooperative Communications2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Wireless communication has been evolving and growing rapidly during the last few years. In order to meet the demand of this growth researchers as well as industry have been working for new techniques and new standardizations. Usually wireless communications performance is badly affected by channel fading. Diversity technique is used to reduce the impact of fading on wireless channel by conveying data over multiple independent fading channel paths and combine them at the receiver. Cooperative diversity is novel transmission technique where multiple terminals use their resources in cooperative manner to form a virtual array that realizes spatial diversity gain in a distributed fashion. Cooperative system can obtain reliable communications, capacity gain and energy saving by mutual relaying among users of wireless network. In this thesis work we examine the performance of a cooperative system and its ability to realize the advantages and the ideas behind its invention. We model a simple decode and forward repetition-based cooperative system consisting of three nodes representing source, destination and relay, which communicate cooperatively between the source and the destination. To evaluate the performance of cooperative system we measure the probability of error of cooperative system using computer simulation. Our results show that a better performance of a communication system could be achieved by allowing wireless network partners to communicate cooperatively. In the last part of the thesis, and for more investigation of the new technique we built a test bed to implement cooperative system model so as to measure its performance practically. Click router and computers with wireless cards are used for the implementation. Click programming language is used to configure the network nodes.

  • 207. Kondabathini, Vishal
    et al.
    Boutamina, Sara
    Vinjarapu, S. K D
    A theme to unite the resources of different remote laboratories2011Conference paper (Refereed)
    Abstract [en]

    Since a decade an extensive research work is being carried out in the field of remote laboratories, which has changed and made the laboratory education system more interesting, cost effective and an efficient usage factor. These labs allow an engineer or a student to practice the experiments they are interested in as many times as they want with a flexibility of place and time. Now we have several remote lab technologies that have evolved, implemented and are working to enhance and simplify the education system of how an engineer/ a student learns his laboratory work. This paper presents a roadmap for universities wanting to deliver engineers with solid documented laboratory experience qualified to design goods and services for a sustainable society. Students need a lot of learning and training covering many aspects of practical engineering and science to become true experimenters. Ubiquitous laboratories with free access and many well-designed experiments are required with the possibility to perform every experiment within the infrastructures 24/7. A challenge is how to make the labs and the collection available for learning and assessment of practical experiments. How should the funding of the laboratories be arranged? Should universities without laboratory resources pay a subscription when using remote laboratories and these databases of experiments at other universities? This paper describes a theme to unite the different resources of remote laboratories which offer the learners/ job seekers the ability to access and use a variety of practical equipments situated in different and distant laboratories through internet without the need to physically moving to these laboratories. In fact this is one of the benefits of using Remote Laboratories.

  • 208. Kulesza, Wlodek
    et al.
    Chen, Jiandan
    Khatibi, Siamak
    Arrangement of a Multi Stereo Visual Sensor System for a Human Activities Space2008In: Stereo Vision / [ed] Bhatti, Asim, Vienna: InTech Education and Publishing , 2008, p. 152-172Chapter in book (Refereed)
    Abstract [en]

    The first part of this chapter introduces a mathematical geometry model which is used to analyze the iso-disparity surface. This model can be used to dynamically adjust the positions, poses and baseline lengths of multiple stereo pairs of cameras in 3D space in order to get sufficient visibility and accuracy for surveillance, tracking and 3D reconstruction. The depth reconstruction accuracy is quantitatively analyzed by the proposed model. The proposed iso-disparity mathematical model presents possibility of reliable control of the iso-disparity curves’ shapes and intervals by applying the systems configuration and target properties. In the second part of this chapter, the key factors affecting the accuracy of 3D reconstruction are analysed. It shows that the convergence angle and target distance influence the depth reconstruction accuracy most significantly. The depth accuracy constraints are implemented in the model to control the stereo pair’s baseline length, position and pose. It guarantees a certain accuracy in the 3D reconstruction. The reconstruction accuracy is verified by a cubic reconstruction method. The optimization is implemented by applying the camera, object and stereo pair constraints into the integer linear programming.

  • 209. Kulesza, Wlodek
    et al.
    Wirandi, Jenny
    Lauber, Alexander
    Modeling of Industrial Measurement Systems Considering the Human Factor2006Conference paper (Refereed)
  • 210. Kusuma, Tubagus Maulana
    et al.
    Caldera, Manora
    Zepernick, Hans-Jürgen
    Utilizing Perceptual Image Quality Metrics for Link Adaptation Based on Region of Interest2005Conference paper (Refereed)
    Abstract [en]

    An implicit link adaptation technique based on hybrid automatic repeat request (H-ARQ) and soft-combining is considered for transmission of Joint Photographic Experts Group 2000 (JPEG2000) images over wireless channels. Adaptation is carried out utilizing an objective perceptual image quality metric that takes into account the human perception. Retransmissions focus on the Region of Interest (ROI) part of the JPEG2000 image to efficiently utilize the bandwidth. Numerical results show that the combination of the proposed perceptual image quality metric with link adaptation provides robust link performance while meeting satisfactory quality constraints.

  • 211. Kusuma, Tubagus Maulana
    et al.
    Zepernick, Hans-Jürgen
    Objective Hybrid Image Quality Metric for In-Service Quality Assessment2005In: Signal Processing for Telecommunications and Multimedia / [ed] Wysocki, Tadeusz; Honary, Bahram; Wysocki, Beata, Berlin: Springer , 2005, p. 44-55Chapter in book (Refereed)
    Abstract [en]

    User-oriented image quality assessment has become a key factor in multimedia communications as a means of monitoring perceptual service quality. However, existing image quality metrics such as Peak Signal-to-Noise Ratio (PSNR) are inappropriate for in-service quality monitoring since they require the original image to be available at the receiver. Although PSNR and others are objective metrics, they are not based on human visual perception and are typically designed to measure the fidelity. On the other hand, the human visual system (HVS) is more sensitive to perceptual quality than fidelity. In order to overcome these problems, we propose a novel objective reduced-reference hybrid image quality metric (RR-HIQM) that accounts for the human visual perception and does not require a reference image at the receiver. This metric is based on the combination of several image artifact measures. The result is a single number, which represents overall image quality.

  • 212. Kusuma, Tubagus Maulana
    et al.
    Zepernick, Hans-Jürgen
    Caldera, Manora
    On the Development of a Reduced-Reference Perceptual Image Quality Metric2005Conference paper (Refereed)
    Abstract [en]

    User-oriented image quality assessment has be- come a key factor in wireless multimedia commu- nications. In particular, perceptual quality assess- ment methods are required to measure the overall perceived service quality based on the grading given by human subjects. This paper focuses on the de- velopment of a reduced-reference perceptual image quality metric, which can be applied for in-service quality monitoring and link adaptation purposes. In contrast to the conventional image ¯delity metrics such as the peak signal-to-noise ratio (PSNR), the proposed hybrid image quality metric (HIQM) takes the human perception into account. In addition, HIQM does not rely on the availability of the full reference of the original image at the receiver.

  • 213. Lagö, Thomas L
    et al.
    Claesson, Ingvar
    Håkansson, Lars
    Overview on Adaptronics and the Future with Smart Embedded Systems2007Conference paper (Other academic)
    Abstract [en]

    Noise and vibration exposure for humans has been of interest for many years. Empirical data has verified that too high dose values can create multiple problems to a human body - often severe. Therefore, spot dose measurements have been used to verify that exposure levels are within acceptable limits. In 2006, the European Machinery Directive increased the responsibility for manufacturers and employers to make sure that limits are in compliance with legislation. The type of classical technology currently used, often employs passive solutions aimed at trying to reduce noise and vibration levels. For low frequency applications, these methods are often inadequate and lack the required level of performance. A smart combination of passive and active techniques can provide many improvements. Today, with possibilities for low cost and embedded electronics, and the rapid development of new actuators, a vast range of applications are possible for this combined approach. A financial advantage also exists when combining active and passive control methods. Today, multiple products utilize embedded "electronic muscles" that counteract harmful sound and vibrations in real-time - featuring advanced control systems. All this then becomes an invisible function in the vehicle, machine, or tool. The launch of such systems and successful acceptance requires international compliance with applicable standards. Via extensive international networks and scientific leadership in multiple organizations, we have managed to develop a range of new products meeting the requirements of international standards. One project, I-SPIDER, is aimed at addressing environmental parameters in schools: an area where our new smart and embedded technology plays a key role. Acticut®, the company that won the Large Embedded Award in Sweden (2006) is another example of how a smart and embedded system can accomplish a task that many people believe to be impossible. New sensor concepts and approaches in combination with consumer and automotive wireless and network solutions are key ingredients for success. It is impossible to imagine how the smart and embedded technology will revolutionize the sound and vibration business and open completely new business segments. This paper will outline these new markets and applications.

  • 214. Lagö, Thomas L
    et al.
    Håkansson, Lars
    Åkesson, Henrik
    Classification of metal cutting vibrations, is it all chatter?2008Conference paper (Other academic)
    Abstract [en]

    Chatter vibrations are causing large monetary losses daily in industry. New materials have increased the challenges with harmful vibration levels. Since the vibrations, when observed as a final result, are chaotic and the vibration process nonlinear, it is a challenging task to deal with it. It is also a common “understanding” in the cutting industry that chatter is RPM (the rotational speed) dependent, since the behaviour changes with RPM. Many attempts have been done over many years to mitigate and understand the vibrations. In our vast research on these topics, we have found that it is rewarding to classify the vibrations into categories, enabling a better understanding of its underlaying physics and “source of vibrations,” and thus also the formulation of a possible remedy. An analysis approach has been developed where vibrations are analyzed and categorized and a GO/NOGO indicator is telling if the machine has the “right type of vibrations.” The work includes a unique solution to inhibit the chatter process and allowing metal cutting without harmful vibrations, the Acticut™ Smart Cutting product line. This article will discuss how such machine testing can be performed and what solutions are at hand, thus saving important money for the companies and also increases quality.

  • 215. Lagö, Thomas L
    et al.
    Zimmergren, Rolf
    Boyer, Alan
    Claesson, Ingvar
    Håkansson, Lars
    New cutting technology provides improved workpiece finish and increased tool life2006In: SOUND AND VIBRATION, ISSN 0022-460X , p. 7-8Article in journal (Refereed)
  • 216. Landqvist, Ronnie
    Signal Processing Techniques in Mobile Communication Systems: Signal Separation, Channel Estimation and Equalization2005Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    Over the last decade there has been an explosive growth in the use of wireless mobile communications. Second generations systems are mature technologies now and third generation systems and beyond are being implemented and researched. Future systems should support a substantially wider and enhanced range of services and thus would require even higher data rates compared to current system in order to deliver these services. A fundamental necessity for being able to provide high data rates is that the physical channel between transmitter and receiver is efficiently utilized. Signal processing algorithms are integral part of any wireless mobile communication systems that makes this possible. In this thesis, several signal processing techniques for improving the performance and capacity of wireless mobile communications systems are discussed. In Part I a simulation package for the simulation of communication systems is implemented and verified. In Part II, the linear and non-linear Projection Approximation Subspace Tracking with Deflation (PASTD) algorithms are proposed for Blind Source Separation (BSS) and Independent Component Analysis (ICA) of linearly mixed signals, respectively. Here, the signals are transmitted simultaneously from multiple antennas. In Part III, the PASTD algorithm is compared to the Rao-Principe (RP) and the Exact Eigendecomposition (EE) algorithm for the purpose of assessing their performance for different configurations. Beamforming is an important function of receivers, in particular to base stations, and in Part IV an efficient and effective space-time adaptive algorithm is proposed. In Part V, an adaptive blind equalization technique for time varying multipath fading channels is suggested and analyzed, and in Part VI a combined channel estimation algorithm for coherent detection in mobile communication systems is proposed. Finally, Part VII investigates several algorithms for power allocation in Multiple-Input Multiple-Output (MIMO) systems and the impact of channel estimation error on the performance of the system is evaluated. In this thesis, the above mentioned algorithms are implemented in Matlab and their applicability and effectiveness were investigated by using several performance measures via Monte Carlo simulation approach. The simulation results clearly demonstrate the promise of using these different signal processing algorithms for improving the performance of wireless mobile communication systems.

  • 217. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    An Efficient and Effective Pilot Space-Time Adaptive Algorithm for Mobile Communication Systems2005In: Radioengineering, ISSN 1210-2512, E-ISSN 1805-9600, Vol. 14, no 1, p. 29-31Article in journal (Refereed)
    Abstract [en]

    In this paper we present a new adaptive space-time algorithm for mitigating the effects of CCI and ISI and minimising the probability of error in mobile communication systems, and evaluate its performance for different mobile velocities. The proposed algorithm is computationally efficient and provides better performance than the conventional RLS algorithm.

  • 218. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    An Efficient and Effective Pilot-Based Space-Time Adaptive Algorithm for Mobile Communication Systems2004Conference paper (Refereed)
    Abstract [en]

    In this paper we present a new adaptive space-time algorithm for mitigating the effects of co-channel interference (CCI) and intersymbol interference (ISI) in mobile communication systems, and evaluate its performance for different array configurations and mobile velocities. The proposed algorithm is computationally efficient and provides better performance than the conventional RLS algorithm.

  • 219. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    Comparative Performance of Three Algorithms for Principal Component Analysis2006In: Radioengineering, ISSN 1210-2512, E-ISSN 1805-9600, Vol. 15, no 4, p. 84-90Article in journal (Refereed)
  • 220. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    EFFECT OF NORMALIZATION OF EIGENVECTORS ON THE PAST AND RP ALGORITHMS FOR PCA2005Conference paper (Refereed)
    Abstract [en]

    This paper investigates the effect of incorporating normali-zation of the eigenvectors between iterations in the PAST and RP algorithms for principal component analysis (PCA). In addition, an algorithm denoted as exact eigendecomposi-tion (EE) is proposed for PCA. The algorithms are compared for different configurations using Monte Carlo simulations. Simulation results show that EE has the best performance and that normalization may be used for improving PAST and RP.

  • 221. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    Novel Application of Projection Approximation Subspace Tracking Algorithm for Whitening in Wireless Communications2005Conference paper (Refereed)
    Abstract [en]

    In this paper we propose two new algorithms that can be used to pre-white signals as a pre-processing step before Independent Component Analysis (ICA). The first algorithm is denoted PASTD for Whitening (PASTD-W) and is based on the Projection Approximation Subspace Tracking with Deflation (PASTD). The second algorithm is a straight forward approach based on exact eigendecomposition, denoted as the Exact Eigendecomposition (EE) algorithm. Computer simulations are performed in order to compare PASTD-W, EE and analytical exact whitening. In these simulations, Binary Phase Shift Keying (BPSK) modulated data are transmitted by two antennas on a channel that acts as linear mixer. The receiver consists of a pre-whitener (PASTD-W, EE or analytical exact whitening) followed by Nonlinear PASTD (NPASTD) for ICA. The simulations show that EE converges faster than PASTD when applied to whitening. Both algorithms show good performance when used together with N-PASTD.

  • 222. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    Simulation of Wireless Digital Communication Systems2004Conference paper (Refereed)
    Abstract [en]

    Due to the explosive demands for high speed wireless services, such as wireless Internet, email and cellular video conferencing, digital wireless communications has become one of the most exciting research topics in electrical and electronic engineering field. In addition, the complexity of wireless communication and signal processing systems has grown considerably during the past decade. Therefore, powerful computer-aided techniques are required for the process of modelling, designing, analyzing and evaluating the performance of digital wireless communication systems. In this paper we present a simple, powerful and efficient way in the simulation of digital wireless communication systems using Matlab. The paper also discusses the basic propagation mechanisms affecting the performance of wireless communication channels. We will also give hints and tips of how to implement these mechanisms in a simulator using Matlab, which can effectively be used for testing and analyzing the performance of different equalization receiver structures and evaluating their effectiveness in combating the destructive effects of the channel.

  • 223. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    Simulation of Wireless Digital Communication Systems2004In: Radioengineering, ISSN 1210-2512, E-ISSN 1805-9600, Vol. 13, no 4, p. 1-7Article in journal (Refereed)
    Abstract [en]

    Due to the explosive demands for high speed wireless services, such as wireless Internet, email and cellular video conferencing, digital wireless communications has become one of the most exciting research topics in electrical and electronic engineering field. The never-ending demand for such personal and multimedia services, however, demands technologies operating at higher data rates and broader bandwidths. In addition, the complexity of wireless communication and signal processing systems has grown considerably during the past decade. Therefore, powerful computer­aided techniques are required for the process of modeling, designing, analyzing and evaluating the performance of digital wireless communication systems. In this paper we discuss the basic propagation mechanisms affecting the performance of wireless communication systems, and present a simple, powerful and efficient way to simulate digital wireless communication systems using Matlab. The simulated results are compared with the theoretical analysis to validate the simulator. The simulator is useful in evaluating the performance of wireless multimedia services and the associated signal processing structures and algorithms for current and next generation wireless mobile communication systems.

  • 224. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    THE IMPACT OF CHANNEL ESTIMATION ERRORS ON THE PERFORMANCE OF MIMO COMMUNICATION SYSTEMS UTILISING WATER FILLING2005Conference paper (Refereed)
    Abstract [en]

    Multipath propagation in wireless channels has traditionally been considered a pitfall that degrades the performance of wireless communication systems. The concept of multiple-input multiple-output (MIMO) channels has turned this view, and today it is evident that users actually benefit from the multipath channel. The MIMO channel promises an information theoretic capacity in bits/s/Hz that grows linearly with the minimum of the number of transmitter and receiver antennas rather than logarithmically as in the case of single-input single-output (SISO) channels. This remarkably finding has led to considerable research in the MIMO-area. In systems where channel state information (CSI) is available, parallel subchannels can be found by Singular-Value Decomposition (SVD) and power can be distributed using the water-filling (WF) procedure. In WF, more power is allocated to "better" subchannels in order to maximize the capacity. Obviously, the resulting performance is dependent of the quality of the CSI estimates. In this paper a MIMO-system simulator for wireless communications is implemented in Matlab. The transmitter and receiver utilize channel estimation and WF in order to maximize capacity. The bit error rate (BER) performance of the system is evaluated in the presence of channel estimation errors of varying degrees.

  • 225. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    The Performance of MIMO Communication Systes Utilising Different Water Filling Strategies2007Conference paper (Refereed)
  • 226. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    The Projection Approximation Subspace Tracking Algorithm Applied to Whitening and Independent Component Analysis in Wireless Communications2005Report (Other academic)
    Abstract [en]

    In Blind Source Separation (BSS) the objective is to extract source signals from their linear mixtures. Algorithms developed for Independent Component Analysis (ICA) have proven useful in the field of BSS. The Projection Approximation Subspace Tracking with Deflation (PASTD) algorithm, originally developed for subspace tracking, has been extended by using a nonlinear cost function so that it may be used for ICA/BSS. Such algorithms most often require the input signals to be white. In this report we extend the PASTD algorithm so that it can be used to whiten signals as a pre-processing step before ICA. The performance of the ICA-algorithm is then evaluated for different choices of whitening algorithms. The algorithms are also evaluated for Binary Phase Shift Keying (BPSK) modulated data over Rayleigh fading channels usually encountered in wireless communications.

  • 227. Larsson, Martin
    Active Noise Control in Ventilation Systems: Practical Implementation Aspects2008Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    In many kinds of buildings, the ventilation is handled by a mechanical ventilation system. Such ventilation systems constitute a well known source of broadband noise. As awareness of the negative effects that subjection to low frequency noise can have on human well-being has increased, so too has the requirement for quieter ventilation installations. Traditionally, duct born noise is attenuated using passive resistive silencers. These passive silencers are valued for their ability to produce a high level of attenuation over a broad frequency range, however they tend to become large and bulky if designed for low frequency attenuation. The active noise control (ANC) technique is known for its ability to produce high levels of attenuation in the low frequency range even with a relatively moderate sized ANC system. On the other hand, ANC normally tends to be ineffective for higher frequencies. Accordingly, a combination of active- and passive techniques, i.e. the construction of a hybrid active/passive silencer, provides a duct silencer solution of manageable size which also covers the low frequency range. The ANC systems controller normally relies on adaptive digital signal processing. Even so, adequate levels of attenuation are not likely to be obtained if the installation of the ANC system is not designed to account for the physical factors that may degrade its performance. This thesis focuses on applying ANC in ventilation systems, with particular emphasis on analysis and installation design, for the purpose of reducing the influence of some of these degrading physical factors. The degrading factors which are of particular interest include: flow induced noise in the microphone signals, acoustic feedback between the control loudspeaker and reference microphone, and standing waves inside the ducts. With respect to installation design, focus is also placed upon industry requests on the ANC system. Taking this into consideration has led to a module based approach, in which the microphones and the loudspeaker are installed in separate modules based on standard duct parts. This thesis comprises four parts. The first describes initial investigations of potential microphone installations intended to reduce flow induced noise. The second part analyzes the influence of flow induced noise on the digital controller and presents further investigations of microphone modules. Further, results of measurements conducted in an acoustic laboratory according to an ISO-standard are presented. The attenuation produced by the ANC system was approximately 15-25 dB between 50-315 Hz even for airflow speeds up to 20 m/s. The third part of this thesis focuses on the possibility of using the passive silencer with which the ANC system is combined, to reduce acoustic feedback and standing waves. The fourth and final part investigates the possibility of using a passive silencer to reduce standing waves in the duct when the ANC system is not installed near the duct outlet.

  • 228. Larsson, Martin
    et al.
    Johansson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    A FEEDFORWARD ACTIVE NOISE CONTROL SYSTEM FOR DUCTS USING A PASSIVE SILENCER TO REDUCE ACOUSTIC FEEDBACK2007Conference paper (Refereed)
    Abstract [en]

    Ventilation systems installed in buildings usually generate low-frequency noise because the passive silencers commonly used to attenuate the ventilation noise are not effective in the low-frequency range. A method proven to effectively reduce low-frequency noise in a wide variety of applications is active noise control (ANC). A feedforward ANC system applied to duct noise normally uses a reference microphone, a control unit, a loudspeaker to generate the secondary noise created by the controller, and an error microphone. The secondary noise generated by the loudspeaker will travel both downstream canceling the primary noise, and upstream to the reference microphone, i.e. acoustic feedback. The acoustic feedback may result in performance reduction and stability problems of the control system. Common approaches to solve the feedback problem result in more complex controller structures and/or system configurations than the simple feedforward controller, e.g. introducing a feedback cancellation filter in the controller in parallel with the acoustic feedback path, or using a dual-microphone reference sensing system. This paper presents a simple approach to reduce the acoustic feedback by using a basic feedforward controller in combination with a passive silencer. Simulations show that efficient acoustic feedback cancellation is achieved by using a passive silencer. In the experimental setup another advantage with using a passive silencer is that the frequency response function of the forward path, which is to be estimated, is smoother, i.e. most of the dominant frequency peaks in the frequency response function when not using a passive silencer is reduced. This in turn results in an acoustic path that is less complex to estimate with high accuracy using an adaptive FIR filter steered with the LMS algorithm.

  • 229. Larsson, Martin
    et al.
    Johansson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    A SYSTEM IMPLEMENTATION OF AN ACTIVE NOISE CONTROL SYSTEM COMBINED WITH PASSIVE SILENCERS FOR IMPROVED NOISE REDUCTION IN DUCTS2007Conference paper (Refereed)
    Abstract [en]

    This paper presents a feedforward active noise control system combined with passive silencers for reducing acoustic noise propagating through ventilation ducts. It is investigated if the passive silencers can increase the noise attenuation potential of the active noise control system and experimental results are presented. The results show that installing a passive silencer results in less pronounced standing waves in the duct and hence to performance increase of the active noise control system. Evaluating measurements regarding the performance of the active noise control system have also been conducted in an acoustic laboratory according to an ISO-standard.

  • 230. Larsson, Martin
    et al.
    Johansson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    Experimental Investigations of Different Microphone Installations for Active Noise Control in Ducts2006Conference paper (Refereed)
    Abstract [en]

    A request on ventilation systems today is the feature of a low noise level. A common method to attenuate ventilation noise is to use passive silencers. However, such silencers are not suitable for the lowest frequencies and one solution is to use active noise control (ANC) to increase the noise attenuation in the low frequency range. Normally when using a feedforward ANC system to attenuate duct noise, both the reference microphone and the error microphone are exposed to airflow. As the airflow excites the diaphragm of the microphones, the microphone signals become contaminated by uncorrelated pressure fluctuations that are not part of the sound propagating in the duct. By reducing the flow velocity around the microphones, these uncorrelated pressure fluctuations can be reduced and the noise reduction improved. One way to reduce the flow velocity around the microphones is to place the microphones in outer microphone boxes connected to the duct via a small slit. In this paper a new practical design for the reduction of flow velocity around the microphones is presented; the microphone installation is based on a T-duct, and therefore it makes maintenance and especially construction easier, compared to the microphone box with a slit. Furthermore, comparative results concerning the performance of an ANC system for the two different microphone installations, the T-duct configurations and the microphone boxes with varying slit width, are presented. The results show that the active noise control performance is almost equal when using the suggested microphone installation as compared to when using a microphone box with a slit.

  • 231. Larsson, Martin
    et al.
    Johansson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    Microphone Windscreens for Turbulent Noise Suppression when Applying Active Noise Control to Ducts2005Conference paper (Refereed)
    Abstract [en]

    Low noise level is an essential feature when installing ventilation systems nowadays. The traditional noise control approaches use passive silencers to attenuate the undesired ventilation noise. These silencers have a high attenuation over a broad frequency range. However, traditional passive silencers are ineffective at low frequencies and tend to be relatively large and bulky when they are used for low frequencies. An approach to improve the low frequency noise attenuation and to reduce the size of a low frequency silencer is active noise control (ANC). A problem when applying ANC to attenuate noise in ducts is that both the reference microphone and the error microphone are placed in an air flow. Accordingly, the microphones sense the sound propagating through the duct as well as the turbulent fluctuations generated by the wind passing over the microphones. The turbulent flow noise reduces the coherence between the reference microphone and the error microphone, resulting in reduced performance of a feedforward ANC system. For improving the performance it is important with as little corruption from turbulent flow noise as possible. The coherence can be improved by reducing the flow velocity around the microphones by using some kind of windscreen. This paper presents comparison results for microphone installations based on different windscreens for suppression of the turbulent wind noise. The presented measurements are carried out in the frequency range 0-400 Hz - the plane wave propagation region for the ducts in use - and for flow speeds up to 5,9 m/s. The results show that with appropriate screens and placement the attenuation and frequency range of attenuation can be significantly improved.

  • 232. Larsson, Martin
    et al.
    Johansson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    Performance Evaluation of a Module Configured Active Silencer for Robust Active Noise Control of Low Frequency Noise in Ducts2008Report (Other academic)
    Abstract [en]

    Low noise level is an essential feature when installing ventilation systems today. Since the passive silencers traditionally used to attenuate ventilation noise tend to become bulky, impractical, and expensive when designed for low frequency attenuation, other solutions for the reduction of the low frequency duct noise often present in ducts are of interest. Active noise control (ANC) is a well known method for attenuating low frequency noise and much research has been performed to successfully apply ANC to duct noise. To insure reliable operation and desirable levels of attenuation when applying ANC to duct noise, it is of highest importance to be able to suppress the contamination of the microphone signals due to the turbulent pressure fluctuations arising as the microphones are exposed to the airflow in the duct. The work presented in this report is concerned with analysis of the influence of the turbulence induced noise on the adaptive algorithm in the ANC system, and design of microphone installations which produce sufficient turbulence suppression while also meeting industrial requirements. These requirements are, for example, that the installations should be based on standard ventilation parts, and that they should be easily installed and maintained. Furthermore, results concerning the performance of an ANC system with different microphone installations are presented. Some of the results were obtained at an acoustic laboratory according to an ISO standard. The attenuation of duct noise achieved with ANC was approximately 15-25 dB between 50-315 Hz, even for airflow speeds up to 20 m/s.

  • 233. Larsson, Martin
    et al.
    Johansson, Sven
    Muddala, S.M.
    Gafar, A.E. Mohamed
    Håkansson, Lars
    Tarkka, Juhani
    Sandor, Mats
    An Initial Study on Applying Active Noise Control to an Insulated Box Fan Used in Ventilation System Applications2009Conference paper (Refereed)
    Abstract [en]

    In many different applications and buildings fans are used to remove unwanted and used air. These fans often generate broadband and tonal noise. Commonly, passive resistive silencers are used to attenuate noise generated by different types of fans installed in ventilation systems. Passive silencers tend to become bulky and impractical when designed for low frequency attenuation. However, active noise control (ANC) is a technique known for its ability to produce high levels of attenuation in the low frequency range, even with a relatively moderate sized ANC system. This paper presents an initial study performed to investigate the possibilities of applying ANC to a radial fan installed inside a box, an insulated box fan. The box is connected to a duct system and can for example be used as a waste air fan. The primary interest in this application, when the fan is used as a waste air fan, is to attenuate the noise generated on the suction side, since that side generates noise into a particular room. Investigations were carried out to determine where the ANC system should be installed, e.g. inside the box, in the duct connected to the box etc. Factors considered were for example, turbulence, standing waves, the type of noise generated by the fan (tonal, broadband, or a combination), and space limitations. The noise generated by the fan was found to be dominated by a tonal component, but also to have broadband energy in the low frequency range. Further, a feedforward ANC system was applied on the suction side, producing approximately 28 dB attenuation of the tonal component, and 5-10 dB attenuation of the broadband noise between 50 and 200 Hz.

  • 234. Lau, Buon Kiong
    et al.
    Cook, G.J.
    Leung, Ye Hong
    AN IMPROVED ARRAY INTERPOLATION APPROACH TO DOA ESTIMATION IN CORRELATED SIGNAL ENVIRONMENTS2004Conference paper (Refereed)
    Abstract [en]

    Many popular direction-of-arrival (DOA) estimators rely on the fact that the array response vector of the array is Vandermonde, for example, that of a uniform linear array (ULA). Array interpolation is a preprocessing technique to transform the array response vector of a planar array of arbitrary geometry to that of a ULA over an angular sector. While good approximation within the target sector is attained in the existing array interpolation approaches, the response of the interpolated array in the out-of-sector region is at best partially controlled. Accordingly, out-of-sector signals, especially those highly correlated with the in-sector signals, can degrade significantly the performance of DOA estimators (e.g., MUSIC with spatial smoothing) that rely on the Vandermonde form to work correctly. In this paper, we propose an improved array interpolation approach that takes into account the array response over the full azimuth. We present also numerical examples to demonstrate the shortcomings of the existing approaches and the effectiveness of our proposal.

  • 235. Lindström, Fredric
    Digital signal processing methods and algorithms for audio conferencing systems2007Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    Today, we are interconnected almost all over the planet. Large multinational companies operate worldwide, but also an increasing number of small and medium sized companies do business overseas. As people travel to meet and do businesses, the already exposed earth is subject to even more strain. Audio conferencing is an attractive alternative to travel, which is becoming more and more appreciated. Audio conferences can of course not replace all types of meetings, but can help companies to cut travel costs and to reduce the environmental strain. The continuously growing market for audio conferencing systems proves that audio conferencing will play an important part in future communication solutions. This thesis treats digital signal processing methods and algorithms for single microphone audio conferencing systems. Concrete real problems, all in relation to audio conferencing systems, are discussed. An intrinsic problem to an audio conferencing system is the acoustic echoes picked up by the microphone. Acoustic echoes are generally cancelled using adaptive fi ltering. In such adaptive filter systems, a major difficulty is to achieve robustness in situations where both participants in a conversation are talking simultaneously. This thesis presents methods and solutions, focusing on the use of parallel adaptive fi lters, which provides the desired robustness. Audio conferencing systems are consumer electronic products and the manufacturing cost is a constant issue. Therefore, it is desirable to implement solutions on low-cost fi nite precision processors. A method to reduce fi nite precision effects in parallel filter implementations is presented in he thesis. In order to run algorithms on low-cost processors it is necessary to keep the computational complexity low. The thesis proposes a number of different methods to reduce complexity,including specific methods targeted for wideband solutions and systems equipped with extension microphones. A high quality audio conferencing system should be equipped with some sort of noise reduction feature. In the end of the thesis a method for integrating such noise reduction with the acoustic echo cancellation is presented. The performance of the proposed methods and algorithms are demonstrated through simulations as well as on real acoustic systems.

  • 236. Lindström, Fredric
    Signal Processing in Audio Conferencing Systems2004Licentiate thesis, comprehensive summary (Other academic)
  • 237. Lindström, Fredric
    et al.
    Borgh, Markus
    Sjöholm, Jonas
    McAllister, Anita
    Södersten, Maria
    Voice and noise: A study in a pre-school environment2009Conference paper (Refereed)
  • 238. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    An open-loop doubletalk detector using power spectrum estimation2004Conference paper (Refereed)
  • 239. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    On audio hands-free design2004Conference paper (Refereed)
    Abstract [en]

    High-quality audio hands-free systems involve rather complex signal processing. The development of such a system is not a straightforward task. This paper proposes a step-by-step approach to the design and implementation of an audio hands-free system. The proposed design method facilitates the implementation process and leads to a robust audio hands-free system solution. The paper also provides an overview of the problems encountered when designing an audio hands-free system. State-of-the-art solutions as well as recently proposed solutions are referred to in addition to the hands-free system design problems.

  • 240. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    The two-path algorithm for line echo cancellation2004Conference paper (Refereed)
    Abstract [en]

    The two-path algorithm is an algorithm for line echo cancellation based on two parallel filters. This paper proposes a modification of the two-path algorithm that improves its performance. In the two-path algorithm a background filter is used for continuously adaptive estimation of the line echo, while a foreground filter is used for the actual cancellation. The coefficients of the background filter are copied into the foreground filter when the background filter is proven to perform better. A robust algorithm for line echo cancellation is thereby achieved. In this paper, the benefits and the drawbacks of the two-path algorithm are evaluated and demonstrated through simulations. A modification is proposed that reduces the negative effects of the two-path algorithm. This modification is compared to the original two-path algorithm. Simulations using real speech signals indicate that the proposed modification can improve the performance of the two-path algorithm. © 2004IEEE.

  • 241. Lindström, Fredric
    et al.
    Eriksson, John-Erik
    Dahl, Mattias
    Claesson, Ingvar
    On the Design of a Sound System for a Mobile Audio Unit2005Conference paper (Refereed)
    Abstract [en]

    A mobile audio unit is a wireless, battery-driven unit, the main purpose of which is to reproduce acoustic signals. This kind of unit can be used in conjunction with a home server. For example, a radio station broadcasting can be received from the Internet and fed to the mobile audio unit via a central home server. The market for home servers is expected to grow leading to a possible expansion of the market for this type of mobile audio unit. This paper presents some design aspects for the sound system of an audio unit, adapted to the new demands of the market.

  • 242. Lindström, Fredric
    et al.
    Schüldt, Christian
    Claesson, Ingvar
    A hybrid acoustic echo canceller and suppressor2007In: Signal Processing, ISSN 0165-1684 , Vol. 87, no 4, p. 739-749Article in journal (Refereed)
    Abstract [en]

    Wideband communication is becoming a desired feature in telephone conferencing systems. This paper proposes a computationally efficient echo suppression control algorithm to be used when increasing the bandwidth of an audio conferencing system, e.g. a conference telephone. The method presented in this paper gives a quality improvement, in the form of increased bandwidth, at a negligible extra computational cost. The increase in bandwidth is obtained through combining a conventional acoustic echo cancellation unit and an acoustic echo suppression unit, i.e. a hybrid echo canceller and suppressor. The proposed solution was implemented in a real-time system. Frequency analysis combined with subjective tests showed that the proposed method extends the bandwidth, while maintaining high quality.

  • 243. Lindström, Fredric
    et al.
    Schüldt, Christian
    Claesson, Ingvar
    An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation2007In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 15, no 4, p. 1320-1326Article in journal (Refereed)
    Abstract [en]

    Adaptive filters for echo cancellation generally need update control schemes to avoid divergence in case of significant disturbances. The two-path algorithm avoids the problem of unnecessary halting of the adaptive filter when the control scheme gives an erroneous output. Versions of this algorithm have previously been presented for echo cancellation. This paper presents a transfer logic which improves the convergence speed of the two-path algorithm for acoustic echo cancellation, while retaining the robustness. Results from simulations show an improved performance, and a fixed-point DSP implementation verifies the performance in real-time

  • 244. Lindström, Fredric
    et al.
    Schüldt, Christian
    Claesson, Ingvar
    Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation2007In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 2007Article in journal (Refereed)
    Abstract [en]

    An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter) for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS)-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.

  • 245. Lindström, Fredric
    et al.
    Schüldt, Christian
    Claesson, Ingvar
    Reusing Data During Speech Pauses in an NLMS-based Acoustic Echo Canceller2006Conference paper (Refereed)
    Abstract [en]

    Fast convergence of the adaptive filter in an acoustic echo cancellation based hands-free communication system is desirable as it implies more periods of possible full-duplex communication. This paper presents a normalized least mean square (NLMS)-based algorithm, targeted for acoustic echo cancellation based units equipped with large external memory. The proposed algorithm utilizes unused processing resources in periods of silence, thus no extra complexity as compared with the conventional NLMS algorithm is required. The improvements obtained by the proposed algorithm are verified through simulated, as well as through real acoustic systems.

  • 246. Lindström, Fredric
    et al.
    Schüldt, Christian
    Dahl, Mattias
    Claesson, Ingvar
    Improving the Performance of a Low-Complexity Doubletalk2005Conference paper (Refereed)
  • 247. Lindström, Fredric
    et al.
    Schüldt, Christian
    Långström, Mikael
    Claesson, Ingvar
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation2007In: IEEE transactions on circuits and systems I-Regular Papers, ISSN 1057-7122, Vol. 54, no 9, p. 2011-2018Article in journal (Refereed)
    Abstract [en]

    The two-path algorithm is an adaptive filter algorithm based on a parallel filter structure, which has been found to be useful for line echo cancellation as well as for acoustic echo cancellation. It is well known that in finite precision arithmetic, the adaptation process of adaptive algorithms can be reduced or even halted due to finite precision effects. This paper proposes a variant of the two-path scheme where the effects of quantization are reduced, without any significant increase in complexity. The improvement is shown by simulations using bandlimited flat spectrum noise as well as real speech signals.

  • 248.
    Loganathan, Srinath
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Adaptation of a Perceptual Video Quality Measure to Low Bitrate Multimedia Applications2005Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    In every engineering approach, technology tries either to match human perfection or to overcome human limitations. Based on this aspect, the present work tries to estimate the behavior of human visual system in the name of video quality. Appraisal of image quality in video and image processing systems plays a vital role in deciding the quality of service, network maintenance and even to compare different service providers. These systems have a wide range of applications from security services till entertainment which includes digital television, internet video, multimedia messaging services e.t.c,.

  • 249. Lundbäck, Jonas
    On Parameter Estimation: Applications in Radio Astronomy and Power Networks2005Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Signal processing is a common part of the modern society used to obtain high functionality in a vast number of applications. As the development of advanced electronics and powerful computers continue, the limit of the functionality in many systems is increased. Furthermore, as the signal processing can be performed on digitalized signals more advanced methods and algorithms can be employed and enhance the results. In radio-based astronomy a new window of discovery against space has opened using digitally based antenna array systems to observe signals with spectral contents ranging up to a major part of the VHF-band. This enables a high degree of flexibility and incorporates several areas e.g. radio astronomy, signal processing and advanced electronic design, in the development and construction of the technology. The usage of digital signal processing is also seen in power networks to control and monitor the state of the system. The power network is a complex and vital construction for the population that demand a high degree of security and reliability. Methods for monitoring and diagnostics are needed. If a fault can be found with high accuracy the time spent on repairs can be kept low reducing the cost and the consumers discontent. This thesis concerns parameter estimation within radio-based astronomy and fault localization on power lines. In this thesis the connection between the two areas is the use of electromagnetic modelling of underlying physical properties, parameter estimation and digitally based equipment used for advanced signal processing. The first area concerns the estimation of properties of electromagnetic waves e.g. direction of arrival and state of polarization, using an antenna array consisting of Tripole antennas. The properties of this antenna and the corresponding array configuration are investigated in part I-III of this thesis. The second area concerns fault localization on power lines using frequency modulated radar techniques. Part IV and V of this thesis present the concept and properties of this fault locator.

  • 250. Lundbäck, Jonas
    On signal processing and electromagnetic modelling: applications in antennas and transmission lines2007Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    This doctoral thesis is comprised of five parts. The first three parts concern signal processing and electromagnetic modelling of multiport antennas. The last two parts concern signal processing and transmission line theory applied to wave splitting on transmission lines. In Part I, the spherical vector wave expansion of the electromagnetic field is used to completely characterize a multiport antenna. A general framework for modelling an antenna configuration based on measurement data and numerical computation is obtained. The generic electromagnetic model for arbitrary multiport antennas or vector sensors is applied in direction of arrival (DOA) estimation. Next, in Part II using the generic electromagnetic model (from Part I), we obtain the Cramér–Rao bound (CRB) for DOA and polarization estimation using arbitrary multiport antennas. In the Gaussian case, the CRB is given in terms of the transmission matrix, the spherical vector harmonics and its spatial derivatives. Numerical examples using an ideal Tripole antenna array and a non-ideal Tetrahedron antenna array are included. In Part III, the theory of optimal experiments is applied to a cylindrical antenna near-field measurement setup. The D-optimal (determinant) formulation using the Fisher information matrix of the multipole coefficients in the spherical wave expansion of the electrical field result in the optimal measurement positions. The estimation of the multipole coefficients and corresponding electric field using the optimal measurement points is studied using numerical examples and singular value analysis. Further, Part IV describes a Digital Directional Coupler (DDC), a device for wave splitting on a transmission line. The DDC is a frequency domain digital wave splitter based on two independent wide-band measurements of the voltage and the current. A calibration of the digital processor is included to account for the particular transmission line and the sensors that are employed. Properties of the DDC are analyzed using the CRB and an experiment where wave splitting was conducted on a coaxial–cable is accounted for. Finally, in Part V the DDC has been designed and implemented for wave splitting on a medium voltage power cable in a power distribution station using low cost wide–band sensors. Partial discharge measurements are conducted on cross–linked polyethylene insulated power cables. The directional separation capabilities of the DDC are visualized and utilized to separate multiple reflections from partial discharges based on the direction of travel.

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