Change search
Refine search result
123456 151 - 200 of 279
CiteExportLink to result list
Permanent link
Cite
Citation style
  • apa
  • harvard1
  • ieee
  • modern-language-association-8th-edition
  • vancouver
  • Other style
More styles
Language
  • de-DE
  • en-GB
  • en-US
  • fi-FI
  • nn-NO
  • nn-NB
  • sv-SE
  • Other locale
More languages
Output format
  • html
  • text
  • asciidoc
  • rtf
Rows per page
  • 5
  • 10
  • 20
  • 50
  • 100
  • 250
Sort
  • Standard (Relevance)
  • Author A-Ö
  • Author Ö-A
  • Title A-Ö
  • Title Ö-A
  • Publication type A-Ö
  • Publication type Ö-A
  • Issued (Oldest first)
  • Issued (Newest first)
  • Created (Oldest first)
  • Created (Newest first)
  • Last updated (Oldest first)
  • Last updated (Newest first)
  • Disputation date (earliest first)
  • Disputation date (latest first)
  • Standard (Relevance)
  • Author A-Ö
  • Author Ö-A
  • Title A-Ö
  • Title Ö-A
  • Publication type A-Ö
  • Publication type Ö-A
  • Issued (Oldest first)
  • Issued (Newest first)
  • Created (Oldest first)
  • Created (Newest first)
  • Last updated (Oldest first)
  • Last updated (Newest first)
  • Disputation date (earliest first)
  • Disputation date (latest first)
Select
The maximal number of hits you can export is 250. When you want to export more records please use the Create feeds function.
  • 151. Lagö, Thomas L
    et al.
    Olsson, Sven
    Various Signal Processing Techniques used on non-Stationary Acoustic Doppler Current Data. Volume I: XI1999Report (Other academic)
    Abstract [en]

    Chapter 1 - Background, deals with the process of analyzing the backscattering signal transmitted from an ultrasonic transducer, [5][28]. The narrowband sinusoidal burst signal is Doppler-shifted due to the current, and this information is converted into current, [14]. The traditional mathematical model for this Doppler process is based on the assumption that the backscattering time signal is Gaussian, due to the Rayleigh backscattering amplitude assumption with random phase, [23][27]. This is based on the assumption that the backscattering is due to many randomly distributed bubbles with about equal size. It is reasonable to question whether this assumption holds for real life signals, [ 1][7][8]. Therefore, this work has concentrated on looking at real life data, and has investigated whether the Gaussian assumption holds for the background noise and the Doppler signal received. It has been found that this is not generally the case. Chapter 2 - Spectral Analysis of Data, provides analysis of the spectral content in the data using tools with different properties. The reason is the difficulty in distinguishing real spectral peaks in the data from peaks coming from variance in the estimate, [2][3]. Therefore, 3D-plots have been generated of current data from four locations around the world with very different environments. Also, a non-linear filtering method named Multiple Peak Count Analysis, MPCA, has been developed. This analysis is most important in understanding if there is more than one Doppler signal component (current) active in the measurement cell analyzed. Using these two methods, which use different foundations for the analysis, it is possible to determine if, and often, how many, Doppler signals are active in one cell. This compares to how many spectral peaks the data contains for each observation interval. Chapter 3 - Statistical Measures provides an analysis of the data using classical statistical tools like histo.gram, normal probability plots, Chi-square tests and variance analysis like ANOVA, ANalysis Of Variance, [2 1][26]. These tools helps in understanding if it is possible to use a Gaussian approach for the signal model, or if some other distribution could be better suited. Data from all four locations are used in the analysis and key results are presented. In this chapter, analysis of the background noise is also analyzed and presented using the above statistical measures. Chapter 4 - Higher Order Moments, provides a description of higher order moments using skewness y, and kurtosis y2. These are important tools of the statistical behavior of the data analysis. The .investigation of the higher order moments for the time series of the three ADCMs, does not contradict the proposed signal model. Furthermore, the real world signals converge very much to what can be expected if this new model is adequate for this kind of signal. The conclusion is, then, that the model holds for this test. The data is found not to obey the Gaussian signal model in general. This is particularly true when the water is troubled. A comparison with real data from four different locations presented above has been performed and all data shows the same trends, the data cannot be modeled using Gaussian statistical properties. The 3D plots presented earlier show that there often are several current vectors active in a cell at the same time, and this has a strong effect on the statistics for the time signal, which is quantified in this chapter. Chapter 5 - Comparison of Estimators, provides an extensive comparison between the covariance method and the Symmiktos MethodTM. Simulated and real data from all four locations have been used in the comparison. The comparison is presented in several formats to make conclusions easier. It is clear that the Symmiktos Method*M generates quite different results from those of the covariance method. On simulated data, the Symmiktos MethodTM is much closer to the simulated truth. However, in real life we don’t know the answer, so it is impossible to be sure which estimator is more accurate. Based on the results from the simulated signals and also noticing that the variance is lower when using the Symmiktos Method M, plus adding the results from the signal model together, it is fairly safe to argue that the Symmiktos Method*M is a more robust and accurate method for Doppler frequency estimation on this type of data. Chapter 6 - Estimator Programs, gives a brief background on the imple-mentation of the main Matlab programs used in the calculations, and the most important programs for understanding of the work, are listed. The programs listed are not only the statistical programs but also the programs used for testing a new signal model and comparing the covariance method with the Symmiktos Method. Chapter 7 - Description of Used Data Sets, gives a brief background on the data sets used in this research. Key data from the four locations is given as well as all the background parameters to when and how the data was collected, as well as the main observations made at the time of data collection. ) Chapter 8 - Summary and Conclusions, provides a summary of the key results from the four different locations. Each method is commented individually and the main effects are discussed. Chapter 9 : References, lists all the references used. Chapter 10 - Listing of Measurement Plots, lists all the plots. The plots consist of about 2000 pages divided over 11 volumes.

  • 152. Lagö, Thomas L
    et al.
    Olsson, Sven
    Eriksson, Per
    An efficient sampling technique applied in an Acoustic Doppler Current Meter1999In: Experimental techniques (Westport, Conn.), ISSN 0732-8818, E-ISSN 1747-1567, Vol. 23, no 6, p. 29-32Article in journal (Refereed)
  • 153. Lagö, Thomas L
    et al.
    Olsson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    Design of an Efficient Chatter Control System For Turning and Boring2002Conference paper (Refereed)
    Abstract [en]

    In the turning operation chatter or vibration is a frequent problem, which affects the result of the machining, and, in particular, the surface finish. Tool life is also influenced by vibration. Severe acoustic noise in the working environment frequently occurs as a result of dynamic motion between the cutting tool and the workpiece. In all cutting operations like turning, boring and milling vibrations are induced due to the deformation of the workpiece. This implies several disadvantages, economical as well as environmental. Many different solutions to minimize the problem have been developed but the fundamental problem is still there. The true nature of the vibrations, its causes, and implications were revealed in a doctor’s thesis in 1999. This has led to a break through in this research area. Since then, through recent research results at Blekinge Institute of Technology, a new approach to controlling vibrations in cutting operations in a lathe has been implemented in a product called ActicutTM, developed by Active Control Sweden AB. This new method controls vibrations in the cutting direction.

  • 154. Lagö, Thomas L
    et al.
    Olsson, Sven
    Håkansson, Lars
    Claesson, Ingvar
    Performance of a Chatter Control System for Turning and Boring Applications2002Conference paper (Refereed)
    Abstract [en]

    In the turning operation chatter or vibration is a frequent problem, which affects the result of the machining, and, in particular, the surface finish. Tool life is also influenced by vibration. Severe acoustic noise in the working environment frequently occurs as a result of dynamic motion between the cutting tool and the workpiece. In all cutting operations like turning, boring and milling vibrations are induced due to the deformation of the workpiece. This implies several disadvantages, economical as well as environmental. Many different solutions to minimize the problem have been developed but the fundamental problem is still there. The true nature of the vibrations, its causes, and implications were revealed in a doctor’s thesis in 1999, [1]. This has led to a break through in this research area. Since then, through recent research results at Blekinge Institute of Technology, a new approach to controlling vibrations in cutting operations in a lathe has been implemented in a product called Acticut®, developed by Active Control Sweden AB. This new method controls vibrations in the cutting direction using embedded sensors and actuators and a filtered-x LMS algorithm. This paper will discuss the application but also the algorithm and its main numerical properties to accomplish a good result, still maintaining its stability properties.

  • 155. Landqvist, Ronnie
    et al.
    Mohammed, Abbas
    An Adaptive Block-Based Eigenvector Equalization for Time-Varying Multipath Fading Channels2005In: Radioengineering, ISSN 1210-2512, E-ISSN 1805-9600, Vol. 14, no 1, p. 20-23Article in journal (Refereed)
    Abstract [en]

    In this paper we present an adaptive block-based eigenvector algorithm (BBEVA) for blind equalization of time-varying multipath fading channels. In addition we assess the performance of the new algorithm for different configurations and compare the results with the least mean squares (LMS) algorithm. The new algorithm is evaluated in terms of intersymbol interference (ISI) suppression, mean squared error (MSE) and by examining the signal constellation at the output of the equalizer. Simulation results show that the BBEVA performs better than the non-blind LMS algorithm.

  • 156. Larson, Lars-Olof
    et al.
    Haan, Jan Mark de
    Claesson, Ingvar
    A New Subband Weight Transform for Delayless Subband Adaptive Filtering Structures2002Conference paper (Refereed)
    Abstract [en]

    In applications like echo cancellation and speech enhancement, where there is need to track changes continuously, adaptive filtering is usually used. Long adaptive filters gives problems like slow convergence and high complexity. Subband adaptive filtering has been introduced to overcome these problems. The filter banks used in subband adaptive filtering introduce large delays. In order to compensate for the delays, delayless subband adaptive filtering is introduced. Delayless subband adaptive filtering is used in both open loop and closed loop configuration, where the subband filters are transformed to a fullband filter using a weight transform. This paper proposes a new subband weight transform based on the filter banks that are used. We investigate the performance of the weight transform using Monte Carlo simulations of a system identification situation. Different adaptive algorithms are used to compare the weight transform to previously proposed weight transforms.

  • 157. Larsson, Sven-Olof
    Capacity Management for Internet Traffic2000Conference paper (Refereed)
    Abstract [en]

    We describe methods to guarantee a certain level of service for Internet traffic by reserving capacity along fixed logical paths. The amount of needed capacity is calculated by using a trade-off between connection rejection probability and the utilisation of the capacity. This can be used in the process of dimensioning capacity on a long term basis. Two dynamic allocation methods are proposed, which periodically reallocates capacity, according to measured traffic loads. Call admission control is used and automatic connection retrials are studied. Each method is designed for a particular scenario. In the first one, a LAN reserves capacity to get a certain transmission speed for every connection. Capacity is paid for according to how much of it that is reserved. The second one considers an aggregated traffic generated by users having limited bandwidth in their connection to the Internet (e.g. modem- and mobile-users) and the operator of the network manages the capacity. The Internet traffic is modeled as Web pa ge fetches from the Worls Wide Web.

  • 158. Larsson, Sven-Olof
    Capacity Management with Logical Links2000Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    To be able to guarantee a certain level of service quality, for different services, one can reserve capacity for these along selected paths through the network. By doing this, one can better control the traffic streams so that the total capacity in the network is used well and so that the services receives the needed quality for the customers. The routing is also simplified in the switches since the path already has been choosen by the use of logical links, which identifies the paths. Different approaches to reallocate capacity between the different trafficstreams has been evaluated and compared with each other. The basic idea is to allocate capacity when it is needed.

  • 159. Larsson, Sven-Olof
    Comparisons of Different Approaches for Capacity Management in ATM Networks2000Conference paper (Refereed)
    Abstract [en]

    The fact that there is a lack of comparisons between methods for virtual path connection (VPC) management makes it important to evaluate fundamentally different methods. We have tried to find representatives of different ways of calculating the capacity distribution. A central approach uses global information about demands and resources, decentralized approaches can be categorized iterative and local. The local approaches that we have evaluated uses the number of ongoing connections to decide how much capacity is needed during the next updating period. An iterative approach uses a distributed way of calculating the capacity distribution. By evaluating these, we have then been able to do a survey over the performances.

  • 160. Larsson, Sven-Olof
    Comparisons of Different Approaches for Capacity Management in ATM Networks2000Conference paper (Refereed)
    Abstract [en]

    The fact that there is a lack of comparisons between methods for VPC management makes it important to evaluate fundamentally different methods. We have tried to find representatives of different ways of calculating the capacity distribution. A central approach uses global information about demands and resources, while decentralized approaches can be categorized into distributed-iterative and local. The local approaches that we have evaluated use the number of ongoing connections to decide how much capacity that is needed during the next updating period. A distributed-iterative approach uses a distributed way of calculating the capacity reservations. By evaluating these, we have then been able to do a survey over the performances.

  • 161. Larsson, Sven-Olof
    Comparisons of Different VPC Capacity Management Methods in ATM Networks2000Conference paper (Refereed)
  • 162. Larsson, Sven-Olof
    et al.
    Arvidsson, Åke
    An Adaptive Local Method for VPC Capacity Management1999Conference paper (Refereed)
    Abstract [en]

    By reserving transmission capacity on a series of links from one node to another within an ATM or SDH/SONET network, making a virtual path connection (VPC) between these nodes, several benefits are obtained. VPCs will simplify routing at transit nodes, connection admission control, and QoS management by traffic segregation. As telecommunication traffic exhibit variations in the number of calls per time unit due to office hours, quick changes in traffic loads (New Year's Eve), and changes in the types of traffic (as in introduction of new services), there is a need to cope for this by adaptive capacity reallocation between different VPCs. We present a type of VPC capacity management policy that uses an allocation function to determine the needed capacity for the coming updating interval based on the current number of active connections. A method for estimating individual VPC congestion states is described together with an adaptive parameter setting of the allocation function.

  • 163. Larsson, Sven-Olof
    et al.
    Arvidsson, Åke
    Different Approaches for Capacity Management1999Conference paper (Refereed)
    Abstract [en]

    The fact that there is a lack of comparisons between methods for VPC management makes it important to evaluate fundamentally different methods. We have tried to find representatives of different ways of calculating the capacity distribution. A central approach uses global information about demands and resources, while decentralized approaches can be categorized into distributed-iterative and local. The local approaches that we have evaluated use the number of ongoing connections to decide how much capacity that is needed during the next updating period. A distributed-iterative approach uses a distributed way of calculating the capacity reservations. By evaluating these, we have then been able to do a survey over the performances.

  • 164. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Computational Efficient Method for Assuring Full Duplex Feeling in Hands-free Communication2003Report (Other academic)
    Abstract [en]

    This report proposes a method for obtaining satisfying "full-duplex feeling" in hands-free communication units at low computational cost. The proposed method uses a combination of an acoustic echo cancellation unit and an adaptive gain unit. The core of the method is to perform the processing of the speech signal into two separate frequency bands and to process these in different manners. Acoustic echoes in the low frequency part of the signal are cancelled by means of an acoustic echo cancellation unit, while acoustic echoes in the high frequency part are suppressed by an adaptive gain unit. The proposed method is well suited when extending the bandwidth of an existing hands-free phone. A real-time implementation of a conventional hands-free phone is compared with a real-time implementation according to the proposed method, where the later is an extended version of the first. The evaluation of the two implementations shows that the proposed method can be used to increase the quality, i.e. extended bandwidth, of a hands-free phone with only a small increase in computational demand.

  • 165. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A computational Efficient Method for Assuring Full-Duplex Felling in Hands-Free Communication2002Conference paper (Refereed)
  • 166. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Computational Efficient Method For Bandwidth Extension of a Conference Phone2003Conference paper (Refereed)
    Abstract [en]

    This paper presents a computationally efficient method for extension of the bandwidth of a conference telephone. The proposed method allows an improvement in quality, i.e. increased bandwidth, at a negligible extra computational cost. This is performed by a combination of an acoustic echo cancellation unit and an adaptive gain unit. The proposed method was implemented in a real-time system. Frequency analysis in combination with subjective tests showed that the proposed method extends the bandwidth with high quality.

  • 167. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Finite Precision LMS Algorithm for Increased Quantization Robustness2003Conference paper (Refereed)
    Abstract [en]

    The well known Least Mean Square (LMS) algorithm, or variations thereof are frequently used in adaptive systems. When the LMS algorithm is implemented in a finite precision environment it suffers from quantization effects. These effects can severely degrade the performance of the algorithm. This paper proposes a modification of the LMS algorithm that reduces the impact of quantization at virtually no extra computational cost. The paper contains an off-line evaluation of a system identification scheme where the presented algorithm outperforms the classical LMS algorithm yielding a better modelling of the unknown plant. This approach is well suited for adaptive system identification, e.g. beam-forming, electrocardiography, and echo cancelling.

  • 168. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Stuns, S
    An LMS Based Algorithm for reduced Finite Precision Effects2002Conference paper (Refereed)
  • 169. Low, Siow Yong
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Robust microphone array using subband adaptive beamformer2002Conference paper (Refereed)
    Abstract [en]

    This paper presents a new robust microphone array to enhance speech signal under the influence of noise and jammer(s). The proposed structure comprises of a soft con-strained subband beamformer, a blocking system and a non-coherent processing technique. The soft constrained beamformer enhances the desired speech signal in a specified region by suppressing all side-lobes. This enhanced signal is then used as a reference signal by the blocking system to estimate the interference spectrum. Finally, the beamformer’s output is spectral subtracted using the estimated interference spectrum. Simulations in a real office environment show higher interference suppression level compared to those obtained using the soft constrained beamformer only. Most importantly, this is achieved with negligible expense on target signal distortion.

  • 170. Low, Siow Yong
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Speech Enhancement Using Multiple Soft Constrained Beamformers and Non-coherent Technique2003Conference paper (Refereed)
    Abstract [en]

    This paper presents a new robust microphone array processing technique to enhance speech signals under the influence of noise and jammer(s). The new structure comprises two soft constrained subband beamformers and a non-coherent processing technique. Essentially, the first beamformer enhances the desired speech signal in a specified constrained region. The residual interference in the beamformer's output is then spectral subtracted using the estimated interference from the second beamformer. Evaluations in a real office environment show higher interference suppression compared to those obtained using the soft constrained beamformer only. Most importantly, this is achieved with negligible expense on target signal distortion.

  • 171. Maillard, Julien
    et al.
    Lagö, Thomas L
    Winberg, Mathias
    Fuller, Chris
    Fluid Wave Actuator for the Active Control of Hydraulic Pulsations in Piping Systems1999Conference paper (Refereed)
    Abstract [en]

    Fluid-borne vibrations in piping systems remain a serious problem in applications such as marine vessels where mechanical fatigue and radiated noise are critical factors. In the case of pumps or hydraulic engines, the main source of vibrational energy is in the fluid axisymmetric plane wave associated with the system pressure pulsations. Due to fluid/structure coupling, this wave propagates in both the pipe wall and fluid. For high levels of pressure pulsations, the resulting radial and axial wall motion can then cause mechanical fatigue and unwanted radiated noise. Passive pulsations dampers have been used traditionally to reduce the fluid pressure pulses. The use of such passive devices is limited however in critical applications due to the resulting static pressure loss which decreases the system performance. This paper describes the design and testing of a non-intrusive fluid wave actuator for the active control of pressure pulses. The actuator consists of a circumferential ring of PZT stacks acting on the pipe outside wall to generate an axisymmetric plane wave in the fluid. Experimental results estimate the control performance of the actuator applied to the discharge line of an oil driven hydraulic engine.

  • 172.
    Martinsson, Daniel
    et al.
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Johnsson, Tobias
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    A Deterministic Channel Model for Simulation of Mobile Radio Communications2002Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    The last decade there has been an enormous expansion in the area of wireless communication. As new services and devices are introduced, and more information is sent between an increasing number of users, more bandwidth is required and the spectrum becomes more limited. To increase capacity in cellular networks, cells can be made smaller and smaller. To be able to plan picocells, such as an indoor environment, in an efficient manner, it is important to have a more detailed understanding about the channel characteristics. Further ways to improve radio communication is to make use of more efficient encoding and receiving techniques, such as spread spectrum. Also when testing new techniques, knowledge about the channel characteristics and limitations are of interest. This thesis models the channel characteristics of an indoor deterministic environment with a simulator using ray tracing techniques. To make the environment as realistic as possible, the physical properties of construction materials are taken into account. The simulator is able to track each individual radio wave, making it possible to calculate interesting parameters such as received power, phase, and direction-of-arrival. The simulator operates in 2D-environments. A lot of work have been done to extend the simulator to a 3D-version, although some problems still remains to be solved.

  • 173. Masud, Mehedi
    et al.
    Das, Gopal Chandra
    Rahman, Anisur
    Ghose, Arunashis
    A hashing technique using separate binary tree2006In: Data Science Journal, ISSN 1683-1470, E-ISSN 1683-1470, Vol. 5, p. 143-161Article in journal (Refereed)
    Abstract [en]

    It is always a major demand to provide efficient retrieving and storing of data and information in a large database system. For this purpose, many file organization techniques have already been developed, and much additional research is still going on. Hashing is one developed technique. In this paper we propose an enhanced hashing technique that uses a hash table combined with a binary tree, searching on the binary representation of a portion the primary key of records that is associated with each index of the hash table. The paper contains numerous examples to describe the technique. The technique shows significant improvements in searching, insertion, and deletion for systems with huge amounts of data. The paper also presents the mathematical analysis of the proposed technique and comparative results.

  • 174.
    Melin, Stefan
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Opto-elektriska korskopplare i optiska nät, Mätningar och nätanalys2002Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [sv]

    Det här examensarbetet som är utfört hos Telia Research AB behandlar opto-elektriska korskopplare (OEXC) i ett modellnät. Arbetet omfattar två delar, en karaktärisering av OEXC:er tillverkade av en specifik leverantör och en nätanalysdel. Karaktärisering av OEXC:er bestod av mätningar utförda i laboratorium på Telia Research AB. Mätningarna syftade till att finna fel och brister hos korskopplaren inför ett fältprov. Mätningar utfördes på fyra OEXC:er. OEXC:erna har 16 portar (16 sändare och 16 mottagare), icke blockerande elektrisk omkopplingsmatris och optiska interface som kan konfigureras för olika bithastigheter. Tester som utfördes var: Konfigurering av noder och länkar, transmissionstester, skyddsomkoppling, tillförlitlighet och stabilitet. Några brister uppdagades vilka rättades till av tillverkaren. Målet med nätanalysdelen var att ta reda på antal portar för korskopplare och antal våglängder som skulle behövas i ett optiskt nät med OEXC:er. Analysen baserades på en nätmodell bestående av två näthalvor med 13 noder i varje halva. Trafikbehovet som låg till grund för analysen baserade på en trafikmatris sombeskriver 2,5 Gbps trafik mellan noder. Transmissionsbegränsningar och ekonomiska aspekter har inte tagits med i analysen. För denna analys antogs tre nätstrukturer; ett stjärnformat nät, ett hierarkiskt nät och ett platt nät. Det stjärnformade nätet har en OEXC per näthalva, det hierarkiska nätet tre OEXC:er per näthalva och det platta nätet har en OEXC per nod. Ur nätanalysdelen kunde man se att de platta nätet krävde mindre än hälften av våglängdsantalet mot det stjärnformade nätet. När det gäller antalet portar på OEXC:er så sjönk det enskilda antalet per OEXC i det platta nätet jämfört med de andra nätstrukturerna. För det platta nätet blir det totala antalet portar fler än dubbelt så många jämfört med det hierarkiska nätet.

  • 175. Mohammed, Abbas
    Advances in Signal Processing for Mobile Communication Systems2002In: International journal of adaptive control and signal processing (Print), ISSN 0890-6327, E-ISSN 1099-1115, Vol. 16, no 8, p. 541-555Article in journal (Refereed)
  • 176. Mohammed, Abbas
    Advances in Signal Processing for Mobile Communication Systems2002In: International journal of adaptive control and signal processing (Print), ISSN 0890-6327, E-ISSN 1099-1115, Vol. 16, no 8, p. 539-540Article in journal (Refereed)
  • 177. Mohammed, Abbas
    Effect of Loran-C Signal Parameters on Skywave Delay Estimation of IFFT Technique2003In: Electronics Letters, ISSN 0013-5194, E-ISSN 1350-911X, Vol. 39, no 14, p. 1091-1093Article in journal (Refereed)
  • 178. Mohammed, Abbas
    Skywave Interference in Loran-C receivers: Causeand Curess1999Conference paper (Refereed)
  • 179. Mohammed, Abbas
    et al.
    Chemnitzer, Holm
    Last, David
    Full Performance Analysis of IFFT Spectral-Division Technique for Skywave Identification in Loran-C2002Conference paper (Refereed)
  • 180. Mohammed, Abbas
    et al.
    Last, David
    Detection and Minimisation of Skywave Interference in Loran-C Receivers1999Conference paper (Refereed)
  • 181. Mohammed, Abbas
    et al.
    last, David
    High Resolution Techniques for Loran-C Skywave Delay Estimation1999In: Electronics Letters, ISSN 0013-5194, E-ISSN 1350-911X, Vol. 35, no 18, p. 1516-1517Article in journal (Refereed)
    Abstract [en]

    Two high-resolution estimation techniques are applied to the problem of estimating the delays of the skywave components of signals input to Loran-C receivers. Their performance is evaluated and compared with that of a Fourier-based spectral-division technique. Simulation results show that the high-resolution algorithms improve the accuracy of the estimates significantly and estimate the skywave delay successfully in situations in which the Fourier-based method fails.

  • 182. Mohammed, Abbas
    et al.
    Last, David
    IFFT Technique for Skywave Detection in Loran-C Receivers2001In: Electronics Letters, ISSN 0013-5194, E-ISSN 1350-911X, Vol. 37, no 6, p. 398-400Article in journal (Refereed)
    Abstract [en]

    An inverse fast Fourier transform (IFFT) technique is presented for isolating groundwave are and skyware components of Loran-C signals in receivers under conditions of very low signal-to-noise ratio. It includes the first example of the measurement of the skywave delays of off-air signals made using this method.

  • 183. Mohammed, Abbas
    et al.
    Last, David
    Novel Signal Processing Techniques for Detecting and Minimizing Skywave Interference in Loran-C Receivers1999In: Navigation, ISSN 0028-1522 , Vol. 46, no 3, p. 147-159Article in journal (Refereed)
    Abstract [en]

    Adaptive skywave estimation techniques that can monitor the delays and strengths of skywave components in real time are desirable in Loran-C receivers since they allow optimal sampling points to be selected. Such techniques should enable the receiver to minimize the errors due to skywave interference while maximizing signal-to-noise and signal-to-interference ratios. This paper reports on novel signal processing techniques employing classical Fourier analysis and modern high-resolution algorithms for identifying skywaves and measuring their delays. The benefits of these methods are assessed, together with the cost in computing resources of implementing them in receivers. The paper includes theoretical analysis, computer simulations, and the results of tests using off-air signals. A prototype Loran-C system employing the proposed techniques is described

  • 184. Mohammed, Abbas
    et al.
    Last, David
    Performance Evaluation of IFFT Technique for Skywave Detection in Loran-C Receivers2000Conference paper (Refereed)
  • 185. Mohammed, Abbas
    et al.
    LeRoux, Fernand
    Last, David
    Eigendecomposition Techniques for Skywave Interference Detection in Loran-C Receivers2003Conference paper (Refereed)
  • 186. Mohammed, Abbas
    et al.
    Samawi, Saad
    Measurement Trials of the Bluetooth Link in Indoor Office Environments2002Conference paper (Refereed)
  • 187. Mårtensson, Björn
    et al.
    Chevul, Stefan
    Järnliden, Håkan
    Johnson, Henric
    Nilsson, Arne A.
    SuxNet – Implementation of Secure Authentication for WLAN2003Report (Other academic)
    Abstract [en]

    Wireless network equipment offers great flexibility for mobile as well as stationary computers. Clients are no longer bound by the length of a network cable. Instead wireless connectivity increases the clients’ mobility. This paper describes an implementation for wireless clients to access a wired computer network through an efficient authentication mechanism. The imple-mentation is called SuxNet, and is a contribution to IP-login [8] and Institute of Electrical and Electronics Engineers (IEEE) 802.1x [3]. The paper also explains and evaluates different security concepts such as Wired Equivalent Privacy (WEP) and IEEE 802.1x.

  • 188.
    Nilsson, Jonas
    et al.
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Nilsson, Jesper
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Low Bitrate Video and Audio Codecs for Internet Communication2003Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    This master thesis discusses the design and the implementation of an own developed wavelet-based codec for both video and image compression. The codec is specifically designed for low bitrate video with minimum complexity for use in online gaming environments. Results indicate that the performance of the codec in many areas equals or even surpasses that of the international JPEG 2000 standard. We believe that it is suitable for any situation where low bitrate is desirable, e.g. video conferences and mobile communications. The game development company Moosehill Productions AB has shown great interest in our codec and its possible applications. We have also implemented an existing audio solution for low bandwidth use.

  • 189. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Digital Filter Design of IIR Filters using Real Valued Genetic Algorithm2003Conference paper (Refereed)
    Abstract [en]

    This paper presents a new paradigm for infinite impulse response (IIR) filter design using genetic algorithms (GA). By encode or transform the filter design problem into the z-plane the GA optimization procedure will be simplified. Additionally, given the z-plane encoding new mutation techniques are introduced, with the intention to locate promising regions in the search space. With proper design of the fitness function, the proposed algorithm can be used to evolve both full precision or quantized filter structures.

  • 190. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    HMM-based speech enhancement applied in non-stationary noise using cepstral features and log-normal approximation2003Conference paper (Refereed)
    Abstract [en]

    This paper proposes a hidden Markov model (HMM)-based speech enhancement method, aiming at reducing non-stationary noise from speech signals. The system is based on the assumption that the speech and the noise are additive and uncorrelated. Cepstral features are used to extract statistical information from both the speech and the noise. A priori statistical information is collected from long training sequences into ergodic hidden Markov models. Given the ergodic models for the speech and the noise a compensated model is created by means of parallel model combination, using a log-normal approximation. During compensation, the mean of every mixture in the speech and noise model is stored. The stored means are then used in the enhancement process to create the most likely speech and noise power spectral distributions using the forward algorithm combined with mixture probability. The distributions are used to generate an optimal linear Wiener filter for every observation. An evaluation of the speech enhancer working in a non-stationary noise environment is performed.

  • 191.
    Nilsson, Mikael
    et al.
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Ejnarsson, Marcus
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Speech Recognition using Hidden Markov Model2002Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    The purpose with this final master degree project was to develop a speech recognition tool, to make the technology accessible. The development includes an extensive study of hidden Markov model, which is currently the state of the art in the field of speech recognition. A speech recognizer is a complex machine developed with the purpose to understand human speech. In real life this speech recognition technology might be used to get a gain in traffic security or facilitate for people with functional disability. The technology can also be applied to many other areas. However in a real environment there exist disturbances that might influence the performance of the speech recognizer. The report includes an performance evaluation in different noise situations, in a car environment. The result shows that the recognition rate varies from 100%, in a noise free environment, to 75% in a more noisy environment.

  • 192.
    Nilsson, Ola
    et al.
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Stjärnborg, Tedh
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Blueboat2002Independent thesis Basic level (degree of Bachelor)Student thesis
    Abstract [en]

    As today (July 2002) there is no doubt that a wireless revolution has arrived. As internet communication is growing stronger, more suitable and advanced techniques are used to integrate offices and home based networks to this new standard. A still somewhat sleeping and awaiting market actor to this new technology is the industrial one. The Blueboat application is meant to replace cables in a military marine system, showing that the wireless solution is as least as good and more flexible as the traditional based on cables. Our task was to send system information such as course, speed and depth from an internal rack to a client computer via Bluetooth. The system information is to be received, interpreted and logged on some kind of software (Bluboat) at the client computer. To do this we are using a microcontroller that is listening on a serial port to receive the information that is to be transmitted. In the microcontroller the data packets are reformatted to Bluetooth packets which are sent to a Bluetooth device connected to the microcontroller On the client side the Bluetooth packets are received and formatted so that it is possible to read real-time data on e.g. a laptop anywhere in the near facility.

  • 193. Nordberg, Jörgen
    Blind Subband Adaptive Equalization2002Report (Other academic)
    Abstract [en]

    It is predicted that a large portion of future wireless communication capacity will be used to provide wireless data services. This new emphasis results in a significant change in substantial parts of the wireless infrastructure, it is important to have low bit error transmission links to get Quality of Service similar to the internet. Future services will also demand higher data rates. These factors mean that there will be a significant need for good quality equalization schemes in the receiver to reduce the different effects of the radio channels, such as intersymbol interference and multipath propagation. The use of linear equalizers leads to high numerical complexity filters even for short delay spread channels. High complexity equalizers implies slow convergence and high numerical load per processed symbol. To save the amount of training or pilot signals and thus increasing the useful data bits, a blind adaptation of the equalizer weights is commonly used. A blind adaptation algorithm in combination with a long equalizer result in even slower convergence. It is therefore desirable to improve the convergence properties of such equalizer schemes. A blind delayless subband equalizer structure is shown to improve the convergence properties of the equalizer. This equalizer consists of a combination of a fullband filtering and and a subband adaptation. In this paper a novel filter bank design and also improved subband-to fullband filter transformation methods is presented. By optimizing the filter bank design, and improving the subband-to-fullband transformation, considerable improvements have been achieved both when it comes to convergence speed and the level of equalization obtained. Simulation results show that the new subband equalization structure has a 12 times faster convergence than its fullband counterpart. The bit error rate is almost the same in both schemes. Thus, there are only advantages by using the suggested equalizer implementation compared to a conventional fullband implementation. The convergence improvement combined with computational savings make it a very attractive technique to use.

  • 194. Nordberg, Jörgen
    Signal Enhancement in Wireless Communications Systems2002Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    Digital Wireless communications has been one of the fastest growing communication techniques during the last decade. Today there exists several different communication systems that use wireless techniques. They share one common property that they transmit data through a radio interface. The radio channel is a tough channel that will both distort and disturb the transmitted signal in various ways. In Jörgen Nordberg's PhD-thesis "Signal Enhancement in Wireless Communications Systems" several different signal enhancement schemes are presented. They have the objective to minimize the impact of the channel. The main part of this thesis presents work on interference cancellation, i.e. how to reduce the impact of other interfering signals on the channel of interest. This is achieved by utilizing the spatial domain, i.e. the receiver is using several antennas to receive the transmitted signals. By using a multitude of antennas techniques like spatial diversity, adaptive antenna arrays, signal separation and beamforming can be applied to combat the interfering signals. In the single antenna case there is often a need to do channel equalization. Since, channel equalization is an inverse filtering, it will often result in estimation of equalization filter parameters of very high order. To reduce the both the complexity and improve the convergence speed of the equalization filter parameter estimation subband processing techniques can be used. In this case the received signal is separated up into different frequency bands (subbands) and decimated according to the bandwidth of the signal. The channel equalization problem is then solved for each subband at a lower sampling rate. Hence, the channel equalization problem is transformed from estimating the parameters of a high order filter into estimating several filter of much lower order.

  • 195. Nordberg, Jörgen
    et al.
    Dam, Hai Huyen
    Nordholm, Sven
    Signal Seperation Using Multi-rate Signal Processing2001Conference paper (Refereed)
    Abstract [en]

    In signal separation and channel equalisation, a short delay spread channel can result in high complexity receiver filters. The computational complexity of estimating these filters can be reduced by using multi-rate technique. In multi-rate processing, a problem is divided into a number of low complexity problems by means of a subband filterbank. Simulation results show up to 93%reduction in computational load compared to a fullband implementation at the expense of a small degradation in separation performance. The results also show that by properly optimizing the filterbank, the performance degradation can significantly be reduced.

  • 196. Nordberg, Jörgen
    et al.
    Dam, Hai Huyen
    Nordholm, Sven
    Subband Signal Separation2002Conference paper (Refereed)
    Abstract [en]

    In wireless communication,the transmitted signals arrive at the receiver will be subjected to multipath propagation,i.e.the signals will be received with different time delays and attenuations.It is therefore not enough to use only a spatial receiver since such a receiver can not exploit the multipath properties. A temporal receiver,on the other hand, works well or multipath situations but does not utilize the spatial information which is of vital importance in multiuser systems.Thus,a combined spatial and temporal receiver structure can utilize multipath,spatial distribution and independence properties of the multiple sources.Such a spatial-temporal receiver structure is usually computationally complex. In this paper a subband signals separator is presented to reduce the required complexity without a significantly reduction in performance. The subband signal separator will be compared with its fullband equivalent using performance measures as the complexity and mean-square-error. Simulation results show that up to 82 % reduction in the computationally complexity with only a slightly degradation in mean square error performance.

  • 197. Nordberg, Jörgen
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Spatial Interference Cancellation using Blind Signal Separation and Sector Antennas2001Conference paper (Refereed)
    Abstract [en]

    In order to increase the capacity of a mobile radio network, it is desirable to exploit the spatial domain in an efficient way. A common technique is to use a sector antennas. Sectors can be formed by using antenna arrays which add spatial domain selectivity in order to improve the Signal-to-Noise and Interference Ratio (SNIR). In this paper an interference cancellation scheme is presented that combines the different areas of beamforming and Blind Signal Separation (BSS). An important feature in the proposed spatial interference canceler scheme is that it does not require any a priori knowledge of the transmitted signals. Consequently there will be no reduction in the system capacity, which is caused by the transmission of training (pilot) signals. Simulation results will show how the different components in the proposed scheme effects the over all performance.

  • 198. Nordberg, Jörgen
    et al.
    Mohammed, Abbas
    Nordholm, Sven
    Claesson, Ingvar
    Fractionally Spaced Spatial Adaptive Equalization of S-UMTS Mobile Terminals2002In: International journal of adaptive control and signal processing (Print), ISSN 0890-6327, E-ISSN 1099-1115, Vol. 16, no 8, p. 541-555Article in journal (Refereed)
    Abstract [en]

    In this paper we present fractionally spaced adaptive equalization techniques and space diversity combined receiver and evaluate their performance for the downlink of S-UMTS system. The conventional "training" (or non-blind) and the "unsupervised" (or blind)adaptive equalization algorithms are both investigated. Simulation results show that the equalizers are robust to Doppler shift and nonlinearity effects due to TWT amplifiers aboard the satellite. It is also shown that even with a moderate array size of two antenna elements, a significant improvement in terminal performance is achieved.

  • 199. Nordberg, Jörgen
    et al.
    Nordholm, Sven
    Grbic, Nedelko
    Mohammed, Abbas
    Claesson, Ingvar
    Performance Improvements for Sector Antennas using Feature Extraction and Spatial Interference Cancellation2002In: IEEE Transactions on Vehicular Technology, ISSN 0018-9545, E-ISSN 1939-9359, Vol. 51, no 6, p. 1685-89Article in journal (Refereed)
    Abstract [en]

    Effective utilization of the spatial domain enhances the capacity of a mobile radio network. A common technique is to use sector antennas, where the sectors are formed by weighting the outputs from the antenna elements. This results in spatial domain selectivity, which significantly improves the signal-to-noise and interference ratio in the received signals. However, the operation of the sector antenna will be limited by the sidelobes of the corresponding beam patterns. By introducing a blind spatial interference canceler that combines the fix beamformers in the sector antenna with blind signal separation, a significant improvement in the multi-user interference suppression can be achieved. Thus, it will be able to efficiently handle the near-far problem, where the users are received with different power. The blind signal separation is performed by an independent component analysis algorithm. The convergence rate of the algorithm is significantly improved compared to the standard formulation by taking into account the modulation format. The algorithm is further improved by introducing a forgetting factor on the weight update. The blind spatial interference canceler is evaluated by simulations using the mean square error and the bit error rate as quality measures. The results show that the mean square error obtained from the blind blind spatial interference canceler is within 0.5 dB from the optimum Wiener solution for signal-to noise ratios greather than 0 dB.

  • 200. Nordebo, Sven
    et al.
    Claesson, Ingvar
    Dahl, Mattias
    Chebyshev Optimization of Circular Arrays2000In: Optimization Methods and Applications, Kluver , 2000Chapter in book (Other academic)
123456 151 - 200 of 279
CiteExportLink to result list
Permanent link
Cite
Citation style
  • apa
  • harvard1
  • ieee
  • modern-language-association-8th-edition
  • vancouver
  • Other style
More styles
Language
  • de-DE
  • en-GB
  • en-US
  • fi-FI
  • nn-NO
  • nn-NB
  • sv-SE
  • Other locale
More languages
Output format
  • html
  • text
  • asciidoc
  • rtf