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  • 351. Shaheem, Asri
    et al.
    Zepernick, Hans-Jürgen
    Caldera, Manora
    Enhanced Channel Shortened Turbo Equalization2008Conference paper (Refereed)
    Abstract [en]

    In this paper, the use of a channel shortening prefilter in conjunction with a maximum a-posteriori probability (MAP) based turbo equalizer is considered. The prefilter shortens the effective channel, thereby reducing the number of equalizer states. As a result of channel shortening, residual intersymbol interference (ISI) appears at the input to the turbo equalizer and the noise becomes colored. To account for the ensuing performance loss, two enhancements to the scheme are proposed. Firstly, a feedback path is used to cancel residual ISI. Secondly, a carefully selected value for the variance of the noise assumed by the MAP-based turbo equalizer is used. Simulations are performed over the highly dispersive Proakis C channel. It is shown that the proposed enhancements give an improvement of approximately 0.65 dB with respect to the unmodified channel shortened turbo equalizer at a bit error rate (BER) of 10E-5.

  • 352. Shaheem, Asri
    et al.
    Zepernick, Hans-Jürgen
    Caldera, Manora
    Prefiltered Turbo Equalization with SINR Mismatch2007Conference paper (Refereed)
    Abstract [en]

    In this paper, we consider the use of a channel shortening prefilter in conjunction with a turbo equalizer, in order to allow its use with arbitrarily long channel impulse responses. We show that the residual intersymbol interference (ISI), caused by imperfect channel shortening, results in considerable performance loss. However, by intentionally introducing a particular signal-to-interference plus noise ratio (SINR) mismatch, some of the penalty incurred can be overcome. We also show that the coloring of the noise through the prefilter results in a significant performance loss, which is insensitive to SINR mismatch and cannot be improved by choosing an appropriate SINR mismatch.

  • 353.
    Silva, Lakmal
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Bo, Zhu
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Blind source separation in real time using Second Order statistics2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    A multi stage approach to speech enhancement methods improves the quality of separation over standard techniques such as spectral subtraction and beamforming. Two algorithms are implemented for convolutive mixtures in two of the important stages of a speech enhancement system, the source separation stage and the background denoising stage. For source separation, a blind source separation method based on second order statistics has been adopted whereas for background denoising, a method based on minimum statistics of subband power, has been used. An efficient real time algorithm for convolutive blind source separation of broad band signals has been realized, by exploiting the second order statistics and non-stationarity of the signals. The real time system is capable of separating sources in a two microphone setup.

  • 354. Sjögren, Thomas
    et al.
    Vu, Viet Thuy
    Pettersson, Mats
    A Comparative Study of the Polar Version with the Subimage Version of Fast Factorized Backprojection in UWB SAR2008Conference paper (Refereed)
    Abstract [en]

    This paper presents a comparative study of the polar and the subimage based variants of the time domain SAR algorithm Fast Factorized Backprojection. The difference between the two variants with regard to the phase error, which causes defocusing in the image, is investigated. The difference between the algorithms in interpolation between stages is also discussed. To investigate the sidelobes in azimuth, the paper gives simulation results for a low frequency UWB SAR system for both algorithms. How the algorithms differ with regard to amount of beams and length of beams is also discussed.

  • 355. Sjögren, Thomas
    et al.
    Vu, Viet Thuy
    Pettersson, Mats
    Moving Target Relative Speed Estimation in the Presence of Strong Stationary Surrounding Using a Single Antenna UWB SAR System2008Conference paper (Refereed)
  • 356. Sjögren, Thomas
    et al.
    Vu, Viet Thuy
    Pettersson, Mats
    Zepernick, Hans-Jürgen
    Gustavsson, Anders
    Speed Estimation Experiments for Ground Moving Targets in Low Frequency UWB2007Conference paper (Refereed)
    Abstract [en]

    This paper presents an iterative method to estimate the Normalised Relative Speed (NRS) of ground moving targets in Ultra Wideband (UWB) wide beam Synthetic Aperture Radar (SAR) using one antenna. The number of iterations depends on the separation between processed NRS and true target NRS. The NRS estimate is based on a chirp rate estimator in azimuth direction of the SAR image. The paper derives an analytical expression of the azimuth phase information based on the moving target NRS and the NRS used in the image formation. The method has been tested on real data from the CARABAS-II SAR system showing good results.

  • 357.
    Sjögren, Tomas
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Vu, Viet
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Pettersson, Mats
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Gustavsson, Anders
    Ulander, Lars
    Moving Target Relative Speed Estimation and Refocusing in Synthetic Aperture Radar Images2012In: IEEE Transactions on Aerospace and Electronic Systems, ISSN 0018-9251, E-ISSN 1557-9603, Vol. 48, no 3, p. 2426-2436Article in journal (Refereed)
    Abstract [en]

    In this paper, a method for moving target relative speed estimation and refocusing based on synthetic aperture radar (SAR) images is derived and tested in simulation and on real data with good results. Furthermore, an approach on how to combine the estimation method with the refocusing method is introduced. The estimation is based on a chirp estimator that operates in the SAR image and the refocusing of the moving target is performed locally using subimages. Focusing of the moving target is achieved in the frequency domain by phase compensation, and therefore makes it even possible to handle large range cell migration in the SAR subimages. The proposed approach is tested in a simulation and also on real ultrawideband (UWB) SAR data with very good results. The estimation method works especially well in connection with low frequency (LF) UWB SAR, where the clutter is well focused and the phase of the smeared moving target signal becomes less distorted. The main limitation of the approach is target accelerations where the distortion increases with the integration time.

  • 358.
    Skoglund, Andreas
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Kader, Risko
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Vidareutveckling av provplattform för mätning av kosmisk strålnings inverkan på DRAM2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [sv]

    Sammanfattning: SAAB Communication i Linköping sysslar med konsultverksamhet mot ett flertal nationella och internationella företag inom både den civila och militära sektorn. Fokus ligger på flyget med uppdrag inom telesystem, radiosystem, signaturanpassning, EMC, atmosfärisk påverkan mm. I det sistnämnda ingår även kosmisk strålnings inverkan på elektronik. Den fortsatta miniatyriseringen av elektronik, speciellt minneselektronik leder till ökad känslighet mot den kosmiska partikelstrålningen som ständigt regnar ner på jorden, därför är det extra noga att minnen testas innan de sätts i bruk vid flygburna system. Syftet med detta examensarbete är att konstruera en provplattform som registrerar fel som uppkommer i ett DDR2 SDRAM minne vid påverkan av den ovannämnda strålningen. Detta examensarbete är en vidareutveckling av en provplattform baserad på en FPGA – lösning. Eftersom denna provplattform saknade stöd för den snabbare minnestypen DDR2 SDRAM, blev syftet med detta examensarbete att ta fram en ny plattform som stödjer denna typ av minne. Den nya provplattformen är en PowerPC baserad lösning från Freescale Semiconductor®. Provplattformen kommer att anslutas till en PC som kör ett program som analyserar antalet detekterade minnesfel, typ av fel samt hur många neutroner minnet har blivit utsatt för under testet. Mjukvaruutvecklingen implementeras i programmeringsspråken assembler samt C. Innan testet av minnet påbörjas, fylls minnet med ett förbestämt bitmönster, om förändringar av bitmönstret sker vid bestrålning av minnet kommer dessa ändringar att registreras tillsammans med den minnesadress där felet inträffade. Abstract: SAAB Communication in Linköping offers consulting toward several national and international companies, both within the civilian and the military market. Focus is in the field of avionics and deals with telecommunication system, radio system, signature adaption, EMC, atmospheric impact etc. The later also includes the influence of cosmic radiation in electronics. The oncoming miniaturisation of electronics, especially within the memory fields causes an increased susceptibility of cosmic radiation that constantly hits our planet, therefore it is of great importance that memories is to be tested before used in airborne systems. The purpose of this thesis is to construct a test platform in order to register faults that occur in a DDR2 SDRAM memory under influence of the mentioned radiation. This thesis is further development of a test platform previously based on a FPGA – solution. Since this test platform had not to support for the more rapidly type of memory as DDR2 SDRAM, the purpose of this thesis became to develop a new platform with support for this type of memory. This new test platform is based on PowerPC from Freescale Semiconductor® The test platform will be connected to a PC running a program that counts the number of detected memory faults, recognises a type of error and records the number of neutrons that the memory has been exposed to during the test. The software is implemented in the programming languages assembler and C. The tested memory will be loaded with a predetermined bit pattern before the test begins, if a change in the bit pattern is detected during exposure of radiation, these faulty bit patterns will be registered together with the memory address where the error has occurred.

  • 359.
    Smedenmark, Daniel
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Search Unit Check: S.U.C2007Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    This thesis addresses the problems in making Non-Destructive Testing of Ultrasonic Unit Checks via a WebService according to the standards ISO TC-135 and SS-EN 12668. The task was to build a WebService which can handle the two standards as well as be able to extract coordinates from a photography on an oscilloscope screen with the corresponding time function. The core problem is to extract the data from a wide range of different types of photographies. The focus in this thesis is on photography with a green component in the graph. Result from the extraction of coordinates shows good accuracy. Only small deviation can be detected if the extracted coordinates are interlaced with the original photography. An runnable WebService has also been developed.

  • 360. Smirnova, Tatiana
    Dynamic Analysis and Modeling of Machine Tool Parts2008Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Boring bar vibration during internal turning operations in machine tools is a pronounced problem in the manufacturing industry. Vibration may easily be induced by the workpiece’s material deformation process, due to the bar’s normally slender geometry. In order to overcome the vibration problem in internal turning active or/and passive control methods may be utilized. The level of success achieved by implementing such methods is directly dependent on the engineer’s knowledge of the dynamic properties of the system to be controlled. This thesis focuses predominantly on three steps in the development of an accurate model of an active boring bar. The first part considers the problem of building an accurate ”3-D” FE model of a standard boring bar used in industry. The influence of the FE model’s mesh density on the accuracy of the estimated spatial dynamic properties is addressed. With respect to the boring bar’s natural frequencies, the FE modeling also considers mass loading effects introduced by accelerometers attached to the boring bar. Experimental modal analysis results from the actual boring bar are used as a reference. The second part discusses analytical and experimental methods for modeling the dynamic properties of a boring bar clamped in a machine tool. For this purpose, Euler-Bernoulli and Timoshenko distributedparameter system models are used to describe the dynamics of the boring bar. Also, ”1-D” FE models with Euler-Bernoulli and Timoshenko beam elements have been developed in accordance with distributedparameter system models. A more complete ”3-D” FE model of the system ”boring bar - clamping house” has also been developed. Spatial dynamic properties of these models are discussed and compared with adequate experimental modal analysis results from the actual boring bar clamped in the machine tool. This section also investigates sensitivity of the spatial dynamic properties of the derived boring bar models to variation in the structural parameters’ values. The final part focuses on the development of a ”3-D” FE model of the system ”boring bar - actuator - clamping house”, with the purpose of simplifying the design procedure of an active boring bar. A linear model is addressed along with a model enabling variable contact between the clamping house and the boring bar with and without Coulomb friction in the contact surfaces. Based on these FE models’ fundamental bending modes, eigenfrequencies and mode shapes, control path frequency response functions are discussed in conjunction with the corresponding quantities estimated for the actual active boring bar.

  • 361. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Dynamic Modeling of a Boring Bar Using Theoretical and Experimental Engineering Methods Part 1: Distributed-Parameter System Modeling and Experimental Modal Analysis2009In: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 14, no 3, p. 124-133Article in journal (Refereed)
    Abstract [en]

    Boring bar vibration is a common problem during internal turning operations and is a major problem for the manufacturing industry. High levels of boring bar vibration generally occur at frequencies related to the first two fundamental bending modes of a boring bar. This is the first of two companion papers that summarize the theoretical and experimental work carried out concerning modeling of dynamic properties of boring bars. This paper introduces the Timoshenko beam theory for the modeling of clamped boring bars. Also, the traditional Euler-Bernoulli beam theory is applied. These continuous system methods have been utilized to produce fixed-free beam models of the clamped boring bar. In order to improve accuracy of dynamic models of clamped boring bars, the modeling of the boring bar clamping is addressed by means of multi-span beam models with pinned boundary conditions. The derived boring bar models have also been compared with results obtained by means of experimental modal analysis, conducted on the actual boring bar clamped in a lathe. The multi-span beam boring bar models display higher correlation with experimental modal analysis results as compared to fixed-free beam models. For the fixed-free beams the Timoshenko model results in the highest correlation with the experimental results. On the other hand, the interval in frequency and the orientation of the two fundamental modes demonstrate differences, particularly between the continuous system models and the experimental results.

  • 362. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Dynamic Modeling of a Boring Bar Using Theoretical and Experimental Engineering Methods Part 2: Finite Element Modeling and Sensitivity Analysis2009In: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 14, no 3, p. 134-142Article in journal (Refereed)
    Abstract [en]

    This is the second of two companion papers that summarize the theoretical and experimental work carried out concerning modeling of dynamic properties of boring bars. This paper introduces the finite element method for the modeling of clamped boring bars. The “3-D” FE models of the system boring bar – clamping house as well as the “1-D” FE models of the clamped boring bar were derived. In particular, the modeling of the boring bar clamping is addressed. Dynamic properties predicted based on the developed FE models of the clamped boring bar were compared with the ones estimated by means of experimental modal analysis conducted on the actual boring bar clamped in the lathe. The “3-D” FE models display substantially higher correlation with the experimental modal analysis results compared to the “1-D” FE models. A “3-D” FE model of the boring bar – clamping house manages to model the distance in frequency and the orientation of the two fundamental modes to a large extent. The importance of the modeling of the boring bar boundary conditions for the accuracy of dynamic models of boring bars is demonstrated. The sensitivity of the natural frequency estimates produced by means of the FE and the continuous system (presented in Part 1) boring bar models with respect to variations in material density and Young’s elastic modulus has been addressed.

  • 363. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Modeling of an Active Boring Bar2007Report (Other academic)
    Abstract [en]

    Vibration problems occurring during internal turning operations in the manufacturing industry urge for adequate passive and/or active control techniques in order to increase the productivity of machine tools. Usually, passive solutions are based on either boring bars made partly in high Young's modulus non-ductile materials such as intered tungsten carbide or boring bars with tuned vibration absorbers adjusted to increase the dynamic stiffness in the frequency range of a certain resonance frequency of the boring bar. By utilizing an active boring bar with an embedded piezoceramic actuator and a suitable controller, the primary boring bar vibrations originating from the material deformation process may be suppressed with actuator-induced secondary "anti-" vibrations. In order to design an active boring bar, several issues have to be addressed, i.e., selecting the characteristics of the actuator, the actuator size, the position of the actuator in the boring bar, etc. This usually implies the manufacturing and testing of several prototypes of an active boring bar, and this is a time-consuming and costly procedure. Therefore, mathematical models of active boring bars incorporating the piezo-electric effect that enable the accurate prediction of their dynamic properties and responses are of great importance. This report addresses the development of a "3-D" finite element model of the system "boring bar-actuator-clamping house". The spatial dynamic properties of the active boring bar, i.e., its natural frequencies and mode shapes, as well as the transfer function between actuator voltage and boring bar acceleration are calculated based on the "3-D" FE model and compared to the corresponding experimentally obtained estimates. Two types of approximations of the Coulomb friction force, the arctangent and the bilinear models, are evaluated concerning modeling contact between the surface of the boring bar and the clamping house.

  • 364. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Accurate FE-modeling of a Boring Bar Correlated with Experimental Modal Analysis2007Conference paper (Refereed)
    Abstract [en]

    In metal cutting the vibration problem of boring bars remains to be one of the most problematic and productivity degrading. A boring bar is very flexible and easily subjected to vibrations due to its large length to diameter ratio, which generally is required to perform internal turning. The boring bar vibrations appear at its first eigenfrequncies, which correspond to the boring bar’s first bending modes that are affected by boring bar’s boundary conditions applied by the clamping and workpiece in the lathe. Therefore the investigation of spatial dynamic properties of boring bars is of great importance for the understanding of the mechanism and nature of boring bars vibrations. This paper addresses the problem of building an accurate 3-D finite element model of a boring bar with ”free-free” boundary conditions. The questions of appropriate meshing and its influence on the boring bar’s spatial dynamic properties estimates as well as modeling the affect of mass loading are discussed. The results from simulations of 3-D finite element model of the boring bar, i.e. its first eigenmodes and eigenfrequencies, are correlated with the results obtained both from experimental modal analysis and analytical calculations using an Euler-Bernoulli model.

  • 365. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Estimation of an active boring bar´s control path FRF:s by means of its 3-D FE-model with Coulomb friction2008Conference paper (Refereed)
    Abstract [en]

    In metal cutting boring bar vibrations may be attenuated using an active boring bar with an embedded piezoceramic actuator, attached error sensor and a suitable controller. In the design of active boring bars accurate modelling of control path frequency response functions (FRF), i.e. FRF between the actuator voltage and the boring bars response signal (which is commonly acceleration), are of importance, e.g. for the decision concerning the favorable position of the actuator inside the active boring bar to maximize vibration suppression. This paper addresses the influence of the Coulomb friction force on the transfer function estimates between the actuator voltage and the boring bar acceleration calculated based on the ”3-D” FE model of an active boring bar. Two types of approximations of the Coulomb friction force, the arctangent and the bilinear models, are evaluated concerning modelling the contact between the surface of the boring bar and the clamping house. Results of incorporation of the two different Coulomb friction force models into the active boring bar’s SDOF model as well as ”3-D” FE-model enabling variable contact between the boring bar, the clamping screws and the clamping house are presented in terms of control path FRF:s.

  • 366. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    INITIAL EXPERIMENTS WITH A FINITE ELEMENT MODEL OF AN ACTIVE BORING BAR2007Conference paper (Refereed)
    Abstract [en]

    One of the most troublesome sources of vibration in metal cutting is the vibration caused by internal turning operations. A boring bar is a tool holder which is used to machine deep precise cavities inside a workpiece material. In order to perform this internal turning the boring bar usually has a large length-to-diameter ratio, and thus the boring bar vibrations are easily excited by the material deformation process during metal cutting. The vibrations are related to the lower order fundamental bending modes of the boring bar. To overcome the vibration problem an active control technique can be used. In particular, by utilizing an active boring bar with an embedded piezoceramic actuator and a suitable controller, the primary boring bar vibrations originating from the material deformation process may be suppressed with secondary "anti-" vibrations. In order to produce an active boring bar several decisions should be done, i.e. the characteristics of the actuator, the position of the actuator in the boring bar, etc. This usually implies that several prototypes of an active boring bar should be produced and tested, thus the design of an active boring bar is a tedious and costly procedure. Therefore a mathematical model which would incorporate the piezo-electric effect in order to predict the dynamic properties and the response of the active boring bar are of great importance. This paper addresses the development of a "3-D" finite element model of the system "boring bar-actuator-clamping house". The spatial dynamic properties of the active boring bar, i.e. its natural frequencies and mode shapes, as well as the transfer function between the voltage applied to the actuator and acceleration of boring bar are calculated based on the "3-D" FE model and compared to experimentally obtained estimates.

  • 367. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Investigation Concerning Actuator Position in an Active Boring Bar Regarding it´s Performance by Means of "3-D" Finite Element Models2008Conference paper (Refereed)
    Abstract [en]

    Active boring bars may be used for active vibration suppression during internal turning operation in metal cutting. This technique is based on a feedback control scheme of the boring bar vibrations measured by an attached sensor (usually accelerometer) where secondary "anti"- vibrations are applied by means of an embedded piezoelectric actuator. In order to design an active boring bar, several issues have to be addressed, i.e. selecting the characteristics of the actuator, the actuator size, the position of the actuator in the boring bar, etc. A mathematical model of the active boring bar incorporating the piezoelectric effect, e.g. a "3-D" finite element, may simplify designing process. In this paper several "3-D" finite element models of the system "boring bar - actuator - clamping house" are developed for a set of actuator positions. The favorable actuator position is basically selected as the one resulting in the greatest "stiffness" of the active boring bar at the frequency corresponding to the first boring bar fundamental bending mode.

  • 368. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Investigation Concerning Dynamic Properties of an Active Boring Bar Regarding its Perfomance by Means of ”1-D” Finite Element Models2009Conference paper (Refereed)
    Abstract [en]

    Active boring bars may be used for active vibration suppression during internal turning operations in metal cutting. This technique is typically based on a feedback control scheme of the boring bar vibrations measured by an attached sensor (usually accelerometer) where secondary ”anti”-vibrations are induced by means of an piezoelectric actuator embedded into a cavity located in the boring bar’s longitudinal direction below its central line. Design procedure of an active boring bar requires the selection of the characteristics of the actuator, the actuator size, the position of the actuator in the boring bar, etc. A ”3-D” finite element model of the active boring bar incorporating the piezoelectric effect was proposed previously to simplify the design process. The set of actuator positions used to decide the favorable actuator position was limited due to time-consuming transient response calculations of the ”3-D” finite element model of an active boring bar. In the present paper a larger set of ”1-D” finite element models of a boring bar (which model the position of the cavity for the actuator but do not incorporate piezoelectric effect) was used to predict dynamic properties of the active boring bar. Based on these results a small set of favorable actuator positions is selected for implementation in ”3-D” finite element models of active boring bars.

  • 369. Smirnova, Tatiana
    et al.
    Åkesson, Henrik
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Simulation of Active Suppression of Boring Bar Vibrations by Means of Boring Bar’s ”1-D” Finite Element Model2009Conference paper (Refereed)
    Abstract [en]

    In metal cutting active control is one method that may be used to attenuate vibration of a boring bar during an internal turning operation. It is based on the utilization of an active boring bar with an embedded piezoceramic actuator and a suitable controller, etc. In this case, the primary boring bar vibrations originating from the material deformation process may be suppressed with secondary "anti-" vibrations induced by the actuator. The design of an active boring bar is usually a tedious and costly procedure, which involves decision making concerning the selection of the actuator characteristics, its position inside the boring bar as well as production and testing of several active boring bar prototypes. Therefore accurate mathematical modeling of the active control system; including the active boring and controller, etc. is of importance. In this paper a simple “1-D” finite element model of a boring bar is utilized to simulate its dynamic response and as controller an adaptive digital controller realized by the feedback filtered-x lms algorithm is used. Control system simulations are presented for the case of broadband excitation.

  • 370.
    Sripada, Praveen
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    MP3 Decoder in Theory and Practice2006Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    MPEG audio coding under the name MP3 has become one of the most popular standards for digital audio broadcasting and videos. High compression ratios offered by MP3 codecs in various stand alone players and hand held devices over the last few years has increased its popularity immensely. Internet users, music lovers who would like to download highly compressed digital audio files at near CD quality are the most benefited. Psychoacoustic model, Modified Discrete Cosine Transform (MDCT) and Huffman coding play a vital role in achieving such magnificent compression ratios. In this thesis, a thorough knowledge of MP3 decoder is obtained by going through the ISO standard and then some of the decoder blocks have been implemented for deeper understanding.

  • 371.
    Svanberg, Niklas
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Gertsovich, Irina
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Analysis of MIG Welding with Aim on Quality2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    Since 1987 Uddcomb Engineering has repaired pulps by their own developed overlay welding method even called Uddcomb method. Currently each welding machine is operated by two persons. To increase Uddcomb Engineering competitiveness the reduced number of operators is desired. An installation of a monitoring system which can aid humans in the welding quality control also helps to improve company’s position. A future goal would be to make this monitoring system automatic without a human operator in the loop. In this thesis, arc voltage, weld current and audio signals were collected and analyzed with aim on finding algorithms to monitor the quality of the welding process. The use of statistics tools is the basis for detecting variations in the voltage and current data, associated with welding process. It has been shown that voltage signal can be used as a part of the welding quality control. The audio signal from welding at low frequencies varies with the speed of the process. The signal can also be incorporated in the monitoring of the process. The use of filters, growing sums and statistics are key elements in the algorithms presented in this report.

  • 372. Swartling, Mikael
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Direction of Arrival Estimation for Multiple Speakers using Time-Frequency Orthogonal Signal Separation2006Conference paper (Refereed)
    Abstract [en]

    This paper presents a new approach for multiple speaker DOA estimation using an array of microphones. The method relies on the fact that multiple independent speakers have a small overlap in the time-frequency domain, i.e. the individual signals are almost W-disjoint orthogonal. By introducing a time-frequency mask and by continuously track the set of time-frequency points corresponding to each individual speech signal, a single source DOA estimation algorithm is used to find the DOA for each separated signal. This approach does not limit the solution to cases where the number of sensors exceeds the number of sources. Real room recordings are used to evaluate the performance of the method where source movements are also included.

  • 373. Swartling, Mikael
    et al.
    Nilsson, Mikael
    Grbic, Nedelko
    Detection of Vehicle Mounted Auditory Reverse Alarm using Hidden Markov Model2007Conference paper (Refereed)
    Abstract [en]

    This paper presents a method for automatically detecting vehicle mounted auditory reverse alarms, or other similar warning signals, based on hidden Markov model and pattern matching techniques. The method is designed for embedded realtime platforms. The purpose of the method is to embed it with active hearing protection devices, aiding the user in detecting warning signals in low SNR environments. Real recordings are used to evaluate the performance, and the results are presented.

  • 374. Swartling, Mikael
    et al.
    Nilsson, Mikael
    Grbic, Nedelko
    Distinguishing True and False Source Locations when Localizing Multiple Concurrent Speech Sources2008Conference paper (Refereed)
    Abstract [en]

    A permutation problem arises in the case of locating multiple speech sources using several sensor arrays in the far field. The intersection of different direction of arrival (DOA) estimates between sensor arrays leads to a set of real source locations as well as a set of false intersections. This paper presents a novel method for pairing DOA estimates from different sensor arrays, resulting in the corresponding real intersection points. The algorithm presented is numerically efficient and suitable for real time implementations. Real room recordings are used to evaluate the method.

  • 375. Swartling, Mikael
    et al.
    Sällberg, Benny
    Grbic, Nedelko
    Direction of Arrival Estimation for Speech Sources using Fourth Order Cross Cumulants2008Conference paper (Refereed)
    Abstract [en]

    In many applications where speech separation and enhancement is of interest, e.g. conferencing systems, mobile phones and hearing aids, accurate speaker localization is important. This paper presents an alternative criteria for the well known Steered Response Power with Phase Transform (SRP-PHAT) algorithm, in which the steered response relates to peaks in the fourth order cross cumulant, rather than peaks in the second order cross cumulant, i.e. the cross power spectrum. Since speech sources have a Probability Density Function (PDF) close to the Laplacian distribution and noise are generally closer to the Gaussian distribution, the fourth order cumulant becomes a good alternative for the steered response search for speech sources. The proposed method is evaluated and compared to the original SRP-PHAT algorithm and shows significant improvements in localization performance for speech sources.

  • 376. Sällberg, Benny
    Applied Methods for Blind Speech Enhancement2008Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    Acoustic disturbances influence human speech communication by interfering with the communication process. In the worst case, it is impossible to communicate at all due to these disturbances. Methods that reduce the influence of the disturbances while preserving speech intelligibility are often desired. This thesis proposes real-world solutions for applied speech enhancement using autonomous and robust methods. Most of the work of the thesis concerns solutions to the problem of reducing acoustic disturbances within the framework of Blind Speech Enhancement (BSE). Notably, the term "blind" is assigned a positive attribute as it implies that the speech enhancement is carried out without any explicit references required. Instead, an assumption about the statistical independence between the sources coupled with an assumption regarding distinguishing statistical properties of the sources underpin the proposed methods. The unifying theory is Independent Component Analysis (ICA), which is performed by means of spatial filtering. Two of the methods that are proposed in this thesis are shown, both in a theoretical and an empirical framework, to be robust in a real application while preserving stability even for Gaussianonly sources. Existing methods cannot guarantee stability in this scenario and Gaussian-only source mixtures may be the case in a real environment. The difference between the two methods lies in the different optimization strategies and the introduced approximations. The idea of injecting a single-channel method into the control loop of a blind beamformer is also proposed. In particular, two approaches are derived that aim at improving the blind beamformer in the case of disturbing noise and maintaining the same performance for different signal input levels. Finally, implementation aspects of a single-channel speech enhancer are discussed. The implementation aspects deal with the implementation of a speech enhancer in several different platforms such as analogue hardware, digital hardware, as well as hybrid analogue and digital hardware.

  • 377. Sällberg, Benny
    Applied Methods to Combat Noise in Human Communication2006Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Acoustic noise has a pronounced negative impact on human vocal communication and user comfort. Interfering noise that is transmitted between communicating users may not only disturb telephone conversation and supporting technical mechanisms (such as speech coders), but may also strain and eventually damage hearing and vocal organs of the users. Furthermore, a user subjected to consistently high levels of noise (such as those emitted by heavy machinery or electrical hand-held tools) may suffer discomfort and health risks. In many cases it is desirable to reduce the level of noise transmitted over a network, as well as controlling and minimizing the level of noise that the local user is subjected to through the utilization of methods for active noise control. This thesis provides applied methods to combat noise in human communication. The active hearing defender is one key application for which the introduced methods may be suitable. The first three parts of this thesis comprise methods for noise reduction in user-to-user vocal communication. In particular, blind methods are investigated. As opposed to conventional (non-blind) methods, blind methods are stand alone and do not require a priori information or knowledge of the spatiotemporal environment. A method based on statistical kurtosis is provided, in which an adaptive, blind subband beamformer is derived and evaluated. The performance of this approach supports speech enhancement in a wide range of applications. A new, low complexity method for blind beamforming is introduced, in which several single channel blind speech enhancers are linearly combined. The nonlinear nature of the blind speech enhancers mean that speech sources add coherently, i.e. performing blind beamforming. Important aspects of implementing a single channel blind speech enhancer in analog, digital and hybrid (mixed analog-digital) hardware are also analyzed. The final section of this thesis outlines an application for active noise control for the purpose of hearing protection. A low-power fixed point digital signal processor is used as an implementation platform as it supports battery powered apparatus. The broad band noise reduction is 20 dB to 30 dB and the tonal interference rejection is 60 dB.

  • 378. Sällberg, Benny
    et al.
    Dahl, Mattias
    Speech Enhancement Implementations in the Digital, Analog, and Hybrid Domain2005Conference paper (Refereed)
    Abstract [en]

    The general quality of speech or important speech parameters such as the intelligibility, clearness or naturalness of speech can be emphasized by signal processing. Such processing for improving speech quality can be found in telecommunication applications, e.g. mobile telephony, internet telephony or personal intercom. By careful selection of domain for realization, i.e. digital, analog, or hybrid, implementation specific benefits can be utilized to increase the speech quality or performance. This paper stresses some key characteristics of the three implementation domains with emphasis on speech enhancement applications. A robust, low complexity, speech enhancement algorithm will be highlighted to illustrate the advantages (and disadvantages) of a purely digital, a purely analog, and a hybrid digital-analog implementation.

  • 379. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    An Adaptive Blind Beamformer with an Integrated Single-Channel Noise Reduction Method for Robust Realtime blind Speech Extraction2008Conference paper (Refereed)
    Abstract [en]

    The performance of single-channel temporal noise reduction methods generally deteriorate in high noise environments, whereas spatial beamformers can maintain some level of speech enhancement. This paper presents a solution where a low complexity single-channel noise reduction method is integrated into the feedback control loop of an adaptive blind beamformer with the purpose of robust blind speech extraction in high noise environments. The proposed combined system outperforms each of the individual methods with respect to signal-to-interference ratio improvement for a wide range of operating conditions, and where the loss in estimated perceptual speech quality due to the combined system is tolerably low. Furthermore, the excess processing load in a hardware solution is comparatively insignificant for the proposed extended approach.

  • 380. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Blind Beamforming Using Parallel Single-channel Speech Enhancers2006Conference paper (Refereed)
    Abstract [en]

    This paper presents an idea to extend a certain class of single channel speech enhancement algorithms to include the spatial domain. The resulting blind beamformer does not rely on a-priori knowledge of source and sensor positions and it enhances one or several speech sources based only on received data. The underlying principle in this approach is the fact that speech signals are short time stationary. Provided that the single channel speech enhancers attenuates unwanted sources and at the same time preserve the short time stationarity of speech signals, a summation of a small array of such single channel processors constitutes a coherent spatial speech enhancement. As opposed to traditional beamforming where the phase alteration is pre-specified, the phase alteration of the proposed structure is controlled by the received data. The evaluation uses a two microphone array and indicates that the Signal to Interference Ratio is increased for a variety of source positions using the proposed method with only an insignificant decrease in speech quality.

  • 381. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Complex-Valued Independent Component Analysis for Online Blind Speech Extraction2008In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 16, no 8, p. 1624-1632Article in journal (Refereed)
    Abstract [en]

    This paper presents a theoretical analysis of a certain criterion for complex-valued independent component analysis (ICA) with a focus on blind speech extraction (BSE) of a spatio–temporally nonstationary speech source. In the paper, the proposed criteria denoted KSICA is related to the well-known FastICA method with the Kurtosis contrast function. The proposed method is shown to share the important fixed-point feature withthe FastICA method, although an improvement with the proposed method is that it does not exhibit the divergent behavior for a mixture of Gaussian-only sources that the FastICA method tends to do, and it shows better performance in online implementations. Compared to the FastICA, the KSICA method provides a 10 dB higher source extraction performance and a 10 dB lower standard deviation in a data batch approach when the data batch size is less than 100 samples. For larger batch sizes, the KSICA metod performs equally well. In an online application with spatially stationary sources the KSICA method provides around 10 dB higher interference suppression, and 1 MOS-unit lower speech distortion compared to the FastICA for 0.15 s time constant in the algorithm update parameter. Thus, the FastICA performance matches the KSICA performance for a time constant above 1 s. Finally, in an online application with a moving speech source, the KSICA method provides 10 dB higher interference suppression, compared to the FastICA for the same algorithm settings. All in all, the proposed KSICA method is shown to be a viable alternative for online BSE of complex-valued signal mixtures.

  • 382. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Implementation Aspects of the Adaptive Gain Equalizer2006Report (Other academic)
    Abstract [en]

    The quality of speech, or important speech parameters such as the intelligibility, clearness or naturalness of speech, can be emphasized by signal processing. Such processing for improving speech quality can be found in telecommunication applications, e.g. mobile telephony, internet telephony or personal intercom. Blind methods are preferable over conventional because they do not require calibration schemes and are independent of environmental variations. By careful selection of hardware domain for realization, i.e. digital, analog, or hybrid, implementation-specific benefits can be utilized to increase the speech quality or performance. This report stresses some implementation aspects when implementing a blind method for speech enhancement in digital, analog, and hybrid digital-analog hardware.

  • 383. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Online Blind Speech Extraction based on a Locally Quadratic Kurtosis Criteria and a Preprocessing Automatic Gain Controller2007Conference paper (Refereed)
    Abstract [en]

    This paper focuses on realtime speech extraction using blind adaptive beamforming. The speech extraction is carried out using an approximation of the kurtosis measure in a subband domain. The introduced kurtosis approximation is an improvement of a recently proposed approximation technique where a locally quadratic criterion was solved at each iteration. The improvement introduced in this paper regards an approach to normalize this same criterion using a pre-processing Automatic Gain Control unit, and thereby making the algorithm invariant to input signal scales. The proposed method outperforms the recent technique in terms of Signal to Interference Ratio improvement. In addition, the increased memory consumption and processing load due to the proposed improvement is comparably low and this is often desirable in a realtime Digital Signal Processor (DSP) implementation. Further, a real-time implementation of the method is conducted and results with real data is presented.

  • 384. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Online Maximization of Subband Kurtosis for Blind Adaptive Beamforming in Realtime Speech Extraction2007Conference paper (Refereed)
    Abstract [en]

    This paper presents a method for blind beamforming with application in realtime speech extraction in a non-stationary environment. The blind beamforming is carried out using an online kurtosis maximization approach where the optimization is based on Newton’s method. The main novelty of the paper lies in the formulation of the subband kurtosis approximation, where a locally quadratic criterion is solved at each iteration. Further, a real-time digital signal processor (DSP) implementation of the method is conducted and results with real data is presented.

  • 385. Sällberg, Benny
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Statistical Analysis of a Local Quadratic Criterion for Blind Speech Extraction2009In: IEEE Signal Processing Letters, ISSN 1070-9908, E-ISSN 1558-2361, Vol. 16, no 2, p. 89-92Article in journal (Refereed)
    Abstract [en]

    This letter aims at complementing previous empirical work regarding a certain beamforming technique for blind speech extraction that uses a local quadratic approximation of a Kurtosis expression. It is shown here that the proposed method possesses a fixed-point property which means that it remains at an optimal solution once this solution has been reached. The proposed method's fixed-point property is valid for a range of source signals including Gaussian sources. This is an improvement over the FastICA method which diverges at the optimal points that correspond to a Gaussian source. In a real application, it cannot be assured that non-Gaussian mixtures are constantly observed; hence, the proposed method is a viable alternative in that case. The fixed-point property further implies that the approximative Kurtosis expression is identical to the true Kurtosis value at an optimal point which, in turn, means that the approximation error is zero. In addition, the convergence towards an optimal solution is always in the direction of a local minimum point even though the optimal solution that correspond to a super-Gaussian source is always a maximum solution which harmonizes with the concept of Kurtosis maximization.

  • 386. Sällberg, Benny
    et al.
    Håkansson, Lars
    Claesson, Ingvar
    Active Noise Control for Hearing Protection using a Lowpower Fixed Point Digital Signal Processor2005Conference paper (Refereed)
    Abstract [en]

    This contribution presents a fixed point implementation for acoustical active noise control in hearing defenders. The well known filtered-x least mean squares structure is conformed to fixed point arithmetic and evaluated in real time. The measured performance of the implementation is 20dB to 30dB attenuation of broad band noise and ca 60dB for sinusoidal interference. The implementation uses a low power fixed point digital signal processor and is well suited for industry application.

  • 387. Sällberg, Benny
    et al.
    Swartling, Mikael
    Grbic, Nedelko
    Claesson, Ingvar
    REAL TIME IMPLEMENTATION OF A BLIND BEAMFORMER FOR SUBBAND SPEECH ENHANCEMENT USING KURTOSIS MAXIMIZATION2006Conference paper (Refereed)
    Abstract [en]

    This paper presents a real time implementation of a blind beamformer for subband speech enhancement. The beamformer adaptively maximizes the statistical kurtosis measure of the beamformer’s output signal. Speech carries high kurtosis and noise often exhibit lower kurtosis. Hence, maximization of the output signal’s kurtosis enhances speech, in general. The implementation is carried out on a novel framework for real time audio processing in MATLAB and uses low latency ASIO sound cards. The implementation is evaluated using recorded signals and the speech is enhanced approximately 10 dB by the proposed approach with perceptually low speech distortion.

  • 388. Sällberg, Benny
    et al.
    Åkesson, Henrik
    Dahl, Mattias
    Claesson, Ingvar
    A Mixed Analog: Digital Hybrid for Speech Enhancement Purposes2005Conference paper (Refereed)
    Abstract [en]

    This paper presents and evaluates a hybrid implementation of a low complexity algorithm for speech enhancement, the Adaptive Gain Equalizer (AGE). The AGE is a subband based time domain method for instantaneous boosting of speech. By combination of digital analysis and analog synthesis, main advantages of the digital domain and the analog domain are utilized. The hybrid solution is implemented on a printed circuit board and evaluated in real-time. The development time of the proposed solution was very short, the solution is flexible, robust, has high signal bandwidth in the signal chain and does not require a Voice Activity Detector (VAD). Furthermore, the solution is not restricted by quantization errors in the signal chain and does not require a high speed Digital Signal Processor (DSP) for analysis. Informal listening tests and Signal-to-Noise Ratio (SNR) measures verify excellent speech enhancement performance and quality.

  • 389. Sällberg, Benny
    et al.
    Åkesson, Henrik
    Westerlund, Nils
    Dahl, Mattias
    Claesson, Ingvar
    Analog Circuit Implementation for Speech Enhancement Purposes2004Conference paper (Refereed)
    Abstract [en]

    Abstract—Human speech is the main method for personal communication. However, interfering noise could degrade the intelligibility of speech, eventually resulting in errors. Thus, efficient speech enhancement algorithms are needed for example in hand held battery powered hearing aids. This paper presents an implementation of a time domain method for speech enhancement purposes; the Adaptive Gain Equalizer. The implementation is carried out on a printed circuit board using common analog electronic components, and evaluated in real-time. The proposed solution advances in prolonged battery life time, high system bandwidth, and it neither quantizes nor digitalizes data as opposed to many digital solutions.

  • 390.
    TAHIR, AUN ALI
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    ZHAO, FENG
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    PERFORMANCE ANALYSIS ON MODULATION TECHNIQUES OF W-CDMA IN MULTIPATH FADING CHANNEL2009Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    The transmission from base station to mobile or downlink transmission using M-ary Quadrature Amplitude modulation (QAM) and Quadrature phase shift keying (QPSK) modulation scheme are consider in W-CDMA system. We can analysis the performance of these modulation techniques when the system is subjected to AWGN and multipath Rayleigh fading are consider in the channel. We will use MatLab 7.4 for simulation and evaluation of BER and SNR for W-CDMA system models. We will go for analysis of Quadrature phase shift key and 16-ary Quadrature Amplitude modulations which are being used in wideband code division multiple access system, so that the system can go for more suitable modulation technique to suit the channel quality, thus we can deliver the optimum and efficient data rate to mobile terminal. Index Terms- AWGN, DSSS, Multipath Rayleigh fading, CDMA, BER, SNR, QPSK, 16- QAM

  • 391.
    Tibblin, Fredrik
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Bengtsson, Tomas
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Aktiv bullerdämpning i ventilationssystem2005Independent thesis Basic level (degree of Bachelor)Student thesis
    Abstract [sv]

    Active headsets have been on the market for years and now it is time for something else to enter the stage. The technology of active noise reduction can for exmple also be used in ventilation systems for reducing ventilation noise. An active control system principally consists of four important components, namely the downstream error microphone, the upstream detection or reference microphone, the digital signal processor (DSP) and a loudspeaker. The upstream reference microphone listen to the primary ventilation noise and sends it to the DSP which makes a phase shift to the signal and sends it to the loudspeaker. The loudspeaker sends out the opposite waveform of the incoming one and reduce the primary ventilation noise. To adjust the sound radrated from the loudspeaker so low reidual soundlevel is obtained a downstream error microphone is used. The residual sound is a sum of the primary ventilation noise and the secondary sound generated by the loudspeaker. In noice reduction systems today there are not many systems that use the active technology. Passive silencers have low noise attenuation at low frequencies, approximately under 250 Hz, and that is why the active technology should be used. The aim with a silencer based on active noise control is to improve the reduction of the low frequency noise below 250 Hz. In this project we also have a passive component that reduce the higher frequencies. The active control system should make the system better in the low frequency range. The use of both active and passive noise reduction at the same time makes an even better system and the goal with this work was to implement a system with high performance to a low cost.

  • 392.
    Tran, Hung
    et al.
    Malardalen Univ, S-72123 Vasteras, Sweden..
    Zepernick, Hans-Juergen
    Blekinge Institute of Technology, Faculty of Computing, Department of Communication Systems. Blekinge Institute of Technology, School of Engineering, Department of Telecommunication Systems. Blekinge Institute of Technology, School of Engineering, Department of Signal Processing. Blekinge Institute of Technology, School of Computing. Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    Phan, Hoc
    Univ Reading, Reading RG6 6AY, Berks, England..
    On Throughput and Quality of Experience in Cognitive Radio Networks2016In: 2016 IEEE WIRELESS COMMUNICATIONS AND NETWORKING CONFERENCE, IEEE , 2016Conference paper (Refereed)
    Abstract [en]

    In this paper, a performance analysis for cognitive radio networks (CRNs) under the outage probability constraint of the primary user and peak transmit power constraint of the secondary user is conducted. Given an automatic repeat request protocol, analytical expressions for the packet delay and throughput of the CRN are derived. Most importantly, these expressions can be used to understand the quality of experience on web services which are assumed to be offered by the considered CRN.

  • 393. Tran, To
    Sequential complex FIR-structure optimization2005Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Over the past 40 years, the area of digital signal processing and filtering has undergone a rapid development. Today, most of the filtering tasks, that before were performed by using analog technology, have been substituted with inexpensive and, often, more reliable and flexible digital solutions. The optimization methods used today for digital filter design date back more than 50 years. Digital filters can mainly be divided into two main categories, Finite Impulse Response (FIR) filters and Infinite Impulse Response (IIR) filters. This thesis deals with the design of three different FIR structures; window functions for frequency analysis, antenna arrays, and a channel equalizer for mobile communication. For all three structures, the corresponding complex optimization problem can become semi-infinite, i.e. a finite number of unknowns with an infinite number of constraints. The Dual Nested Complex Approximation (DNCA) algorithm has been preferred to solve such complex optimization problems. This is partly motivated by the low computational cost and low memory consumption, which allows execution on any desktop or laptop computer. The thesis consists of three parts; Part I considers the design and enhancement of flattop windows constructed with a summation of shifted Dirichlet kernels; Part II deals with the design of antenna arrays, where the antenna element weights are complex-valued; and Part III, finally, employs semi-infinite quadratic programming to design a channel equalizer with a complex-valued response, i.e. with a non-linear phase.

  • 394. Tran, To
    et al.
    Claesson, Ingvar
    Dahl, Mattias
    Design and Improvement of Flattop Windows with Semi-Infinite Optimization2004Conference paper (Refereed)
    Abstract [en]

    Digital and analog window optimization problems are often characterized by a few number of variables with many constraints. In some cases the optimization problem becomes semi-infinite, i.e. a finite number of variables with an infinite set of constraints. This paper presents a method for flattop window design and enhancement using the Dual Nested Complex Approximation (DNCA) algorithm. Flattop windows can be used for accurate amplitude measurements in spectral analysis and can also be used to design FIR filters with very high stopband attenuation. This paper proposes using the DNCA scheme to solve the optimization problem, due to its low computational complexity and memory consumption. It can be run on any desktop computer. The framework of the DNCA scheme is presented together with two examples; one concerning the design of an enhanced version of the ISO 18431-2 flattop window and the other concerns the design of a flattop window which is comparable with the commercial P-401.

  • 395. Tran, To
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Lagö, Thomas L
    High accuracy windows for today's 24bit ADC's2004Conference paper (Refereed)
    Abstract [en]

    This paper presents a new high accuracy window design method based on semi-infinite linear programming by using the Dual Nestled Complex Approximation (DNCA) algorithm. The paper presents a general window design specification, the formulation of the corresponding semi-infinite linear programming solution and several highly optimized window design examples. The paper illustrates the capability to enhance the sidelobe attenuation of the new upcoming ISO 18431-2 flattop window standard from the international organization for standardization. The procedure of enhancing the ISO flattop window can be directly applicable to any other existing windows. Flattop windows are commonly available in frequency analyzers for accurate amplitude measurements of harmonic tones in presence of noise. Moreover, it is possible to design high accuracy windows with more than 150dB sidelobe attenuation with the proposed method as well. Such high sidelobe attenuation is for example needed in today's 24bit equivalent 1bit-Sigma-Delta ADC:s, where high frequency noise is generated.

  • 396. Tran, To
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Lagö, Thomas L
    Window Design and Enhancement using Chebyshev2004Conference paper (Refereed)
    Abstract [en]

    This paper presents a new and versatile framework to window design based on a semi-infinite linear programming approach by using the Dual Nestled Complex Approximation (DNCA) algorithm. The paper considers a practical problem formulation, the general window design specification and the corresponding optimum solution. The DNCA linear programming algorithm is presented and several highly optimized window design examples included. Furthermore, the capability of the design method by enhancing the sidelobe attenuation of existing windows such as Flattop windows is illustrated as well. Flattop windows are commonly available in frequency analyzers for accurate amplitude measurements of harmonic tones in presence of noise. However, the design method can be directly applied to any other existing window, in purpose to enhance its performance. With the proposed method it is possible to design windows with more than 150dB sidelobe attenuation. A relevant application where such high precision windows is needed are in today's 24bit equivalent 1bit-Sigma-Delta ADC:s, where high frequency noise is generated.

  • 397.
    Törnqvist, Henrik
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Bildbehandling för IR2006Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    An infrared video processing simulator has been implemented in MATLAB to support the development and hardware implementation of image processing algorithms in infrared camera systems. The simulator supports functions like non-uniformity correction, dead-pixel replacement, noise filtering, contrast enhancement and output pixel mapping. Five different methods for image stabilization have been evaluated.

  • 398.
    Uzoechi, Victor
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Osigwe, Kenneth
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Fuzzy Modeling of Uplink Transmit Power Control in a CDMA Network2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    From its beginning, transmit power has always placed a significant constraint on the performance of wireless radio systems. The transmit power control problem can be characterized as that of maintaining adequate power in each transmitted waveform so as to increase the expectation that the minimum required SIR at the receiver will at least be reached. This has been shown not to be a trivial endeavor due to the variability of the physical channel with time as well as interference and other practical constraints on “infinitely” increasing transmit power. Several power control algorithms have been proposed, of which the class of distributed and autonomous transmit power control algorithms have been shown in literature to perform quite satisfactorily when compared to centralized schemes due to the moderate complexity that is achievable; and the vast control and signaling overhead that is saved. This thesis work explores the application of fuzzy control to the subject of modeling uplink transmit power control in code division multiple access system. A possible implementation scenario of an SIR-based fully distributed constrained transmit power control algorithm in a multiservice network by applying fuzzy proportional-plus-integral control with a two-input (error and error change) and one-output (transmit power adjustment command) fuzzy rule base and inference engine is proposed.

  • 399.
    Vu, Viet Thuy
    et al.
    Blekinge Institute of Technology, Faculty of Engineering, Department of Mathematics and Natural Sciences. Blekinge Institute of Technology, School of Engineering, Department of Signal Processing. Blekinge Institute of Technology, School of Engineering, Department of Mathematics and Natural Sciences. Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    Pettersson, Mats
    Blekinge Institute of Technology, Faculty of Engineering, Department of Mathematics and Natural Sciences.
    thomas, Sjögren
    Swedish Defence Research Agency, SWE.
    Moving Target Focusing in SAR Image with Known Normalized Relative Speed2017In: IEEE Transactions on Aerospace and Electronic Systems, ISSN 0018-9251, E-ISSN 1557-9603, Vol. 53, no 2, p. 854-861Article in journal (Refereed)
    Abstract [en]

    This paper presents the moving target focusing method, whichallows focusing moving targets in complex synthetic aperture radar(SAR) images without raw data. The method is developed on the rangemigration algorithm, where focusing moving target is an interpolationstep in the wave domain. The simulated results are provided in thepaper to illustrate the proposed method whereas the experimentalresults show its practicality. The method can be flexibly applied fromsmall area to the whole SAR scene.

  • 400. Vu, Viet Thuy
    et al.
    Sjögren, Thomas
    Pettersson, Mats
    A comparison between Fast Factorized Backprojection and frequency-domain algorithms in UWB low frequency SAR2008Conference paper (Refereed)
    Abstract [en]

    Two frequency-domain algorithms Chirp Scaling (CS) with the advantage of simplification and Range Migration (RM) with the advantage of accuracy are candidates for a comparative study to the time-domain algorithm Fast Factorized Backprojection (FFBP) with reference to a UWB system are presented in this paper. The comparison is based on UWB SAR image quality measurements such as spatial resolution, Integrated Sidelobe Ratio (ISLR), Peak Sidelobe Ratio (PSLR) and processing time connected to computational cost. The simulated SAR data, which is used in this study, is based on the parameters of the airborne UWB low frequency CARABAS-II system.

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