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• 251. Tao, Xiao-Jiao
Narrowband acoustic doppler volume backscattering signal-part I: evolutionary spectral analysis2002In: IEEE Transactions on Signal Processing, ISSN 1053-587X, E-ISSN 1941-0476, Vol. 50, no 11, p. 2644-2655Article in journal (Refereed)

This paper examines the statistical properties of the narrowband Doppler volume backscattering process and analyzes its evolutionary spectrum. After clarifying the mechanism of both the finite duration Doppler effect and the continuously space-shifted integration process, the first two order time-varying statistics under a more general assumption, i.e., von Mises distribution, of random phase are derived. The generalization permits nonuniform phase tendency, which occurs in layered medium scattering. Based on the locally stationary process model, the evolutionary spectrum of the signal is derived. It is shown that the variation of the backscattering strength enters the spectrum as an amplitude modulation, whereas the variation of the random phase distribution acts as both the amplitude modulation and the frequency modulation. Finally, the observability of the average flow speed using spectral centroid estimate is discussed.

• 252. Tao, Xiao-Jiao
Narrowband acoustic doppler volume backscattering signal-part II: spectral centroid estimation2002In: IEEE Transactions on Signal Processing, ISSN 1053-587X, E-ISSN 1941-0476, Vol. 50, no 10, p. 2656-2660Article in journal (Refereed)

This paper proposes a novel algorithm to estimate the time-varying spectral centroid of the narrowband Doppler volume backscattering signal. It is constructed in a semi-parametric way, that is, modeling parametrically the local narrowband evolutionary spectrum using an AR(2) and nonparametrically adapting its time-varying coefficients using the wavelet shrinkage. The improved performance is gained by the underlying linear prediction function of the AR(2) and the minimax optimality of the unknown smoothness adaptation of the wavelet shrinkage procedure.

• 253. Tao, Xiao-Jiao
Behind Acoustic Doppler Current Profiler: Narrowband Volume Backscattering Signal Model, Analysis and Estimation1999Report (Other academic)

This report examines the statistical properties of the narrowband Doppler volume backscattering process and discusses its spectral centroid estimation problem. After clarifying the mechanism of both the finite duration Doppler effect and the continuously space-shifted integration process, the first two order time-varying statistics under a more general assumption, i.e. von Mises distribution, of random phase are derived. The generalization permits nonuniform phase tendency, which occurs in layered medium scattering. Based on the locally stationary process model, the evolutionary spectrum of the signal is derived where the variation of the backscattering strength enters as an amplitude modulation. On the contrary, the variation of the random phase distribution acts as both the amplitude modulation and the frequency modulation. The observability and the estimation of the average flow speed are discussed. It is shown that even under the homogeneous condition, the spectral centroid does not coincide with the aver age flow speed. Finally, a semiparametric spectral centroid estimation method, which simply contains an AR(2) model with its time-varying coefficients adapted by a wavelet shrinkage is proposed.

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• 254. Tran, To
A Semi-Infinite Quadratic Programming Algorithm with Applications to Channel Equalization2003Conference paper (Refereed)

This paper is based on DNCA (Dual Nested Complex Approximation) for optimizing communication channel equalizers using semi--infinite quadratic programming. The optimality criterion for the equalizer is either to minimize the complex deviation in the passband or to minimize its stopband energy when subjected to a specified peak side lobe level in the stopband. Additional linear constraints can be used to form the response by means of group delay, nulls etc. The design approach is applied to a numerical example which deals with the design of a complex communication channel equalizer.

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• 255. Westerlund, Nils
Applied Speech Enhancement for Personal Communication2003Licentiate thesis, comprehensive summary (Other academic)

In our daily life noise is almost constantly present. At the same time our need to communicate reliably has probably never been this great. Surrounding noise impairs our ability to communicate by speech and causes the users of a speech communication system to strain both their hearing and their voices. This licentiate thesis deals with speech enhancement for personal communication. A method for speech enhancement mainly aiming at every day communication situations is presented. The method boosts speech energy in a communication system, leaving background noise unaffected and it also tracks changes in background noise characteristics. Hence it is an \emph{adaptive} speech enhancement algorithm. This is desirable since the noise characteristics of most every day noisy situations are more or less rapidly changing. Methods for facilitating personal communication in severely disturbed environments are also presented. Instead of placing the communication microphone in front of the mouth, the microphone is placed inside the external auditory canal of the user. A pair of ear-muffs equipped with an active noise cancelling (ANC) system are fitted onto the user's head. This setup, possibly combined with some speech enhancement method, enables the user to communicate even under the most extreme noise situations. The setup characteristics are examined and the resulting speech quality and intelligibility are evaluated using a speech recognizer based on Hidden Markov Models.

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• 256.
Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
Equalization of Audio Channels: A Practical Approach for Speech Communication2000Independent thesis Advanced level (degree of Master (One Year))Student thesis

Many occupations of today requires the usage of personal preservative equipment such as a mask to protect the employee from dangerous substances or the usage of a pair of ear-muffs to damp high sound pressure levels. The goal of this Master thesis is to investigate the possibility of placing a microphone for communication purposes inside such a preservative mask as well as the possibility of placing the microphone inside a persons auditory meatus and perform a digital channel equalization on the speech path in question in order to enhance the speech intelligibility. Subjective listening tests indicates that the speech quality and intelligibility can be considerably improved using some of the methods described in this thesis.

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• 257. Westerlund, Nils

This report presents a noise reduction method for speech communication where the input signal is divided into a number of subbands that are individually weighted in time domain according to the short time Signal-to-Noise Ratio estimate (SNR) in each subband at every time instant. Instead of focusing on suppression the noise, the method is focusing on speech enhancement. The method has proven to be advantageous since it offers low complexity, low delay and low distortion. Also, there is no need for a Voice Activity Detector (VAD). The method is stand-alone and works regardless of speech coding schemes and other surrounding adaptive systems.

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• 258. Westerlund, Nils
In-Ear Microphone Equalization Exploiting an Active Noise Control2001Conference paper (Refereed)

A pair of ear-muffs that employs active noise control(ANC) for noise reduction, substantially reduces the influence of the low frequencies inside the cap. This implies an indirect high-pass filtering of the sound in the external auditory canal (EAC). This paper shows that the above mentioned high-pass filtering property is convenient when combining an ANC headset with an in-ear microphone (ear-mic) for communication purposes since the speech signal inside the EAC is a low-pass filtered version of the speech signal at the mouth. Hence, the in-ear speech signal is to some extent restored by the ANC high-pass filtering, the quality of the speech signal in the auditory canal is improved and the speech intelligibility is increased. The equalization will also decrease the demand of dynamic range and resolution of electronics used. By that, combining an active ear-muff with an ear-mic serves two purposes: Protecting the user from harmful noise and enables the user to communicate over some channel using the speech signal in the auditory canal.

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• 259. Westerlund, Nils
In-Ear Microphone Hybrid Speech Enhancement2002Conference paper (Refereed)

This paper presents a novel speech enhancement approach for performing noise reduction in severely disturbed environments. A small microphone for communication purposes is placed inside the external auditory canal to pick up the speech signal originating from the speech production organ. The speech enhancement is achieved by using three different noise reduction methods: High frequencies are attenuated by passive absorbers, low frequency components are attenuated by employing active noise control and finally a broadband noise reduction is achieved by using spectral subtraction.

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• 260. Westerlund, Nils
Real-Time Implementation of an Adaptive Gain Equalizer for Speech Enhancement Purposes2003Conference paper (Refereed)

This paper presents a real time low-complexity implementation of a noise reduction method for speech communication where the input signal is divided into a number of subbands that are individually weighted in time domain according to the short timeS ignal-to-Noise Ratio estimate (SNR) in each subband at every time instant. Instead of focusing on suppression the noise, the method is focusing on speech enhancement. The method has proven to be advantageous when implemented in real time since it offers low complexity, low delay and low distortion.

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• 261. Westerlund, Nils
Speech Enhancement using an Adaptive Gain Equalizer2003Conference paper (Refereed)

This paper presents a noise reduction method forspeech communication where the input signal is divided into a number of subbands that are individually weighted in time domain according to the short time Signal-to-Noise Ratio estimate (SNR) in each subband at every time instant. Instead of focusing on suppression the noise, the method is focusing on speech enhancement. The method has proven to be advantageous since it offers low complexity, low delay and low distortion. Also, there is no need for a Voice Activity Detector (VAD). The method is stand-alone and works regardless of speech coding schemes and other surrounding adaptive systems.

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• 262. Westerlund, Nils
Speech Recognition in Severely Disturbed Environments Combining Ear-Mic and Active Noise Control2002Conference paper (Refereed)

A microphone for personal communication is placed inside the external auditory canal of a user. A pair of ear-muffs equippedwith Active Noise Control (ANC) are fitted onto the head of the user. The combination of passive and active noise reduction makes communication more reliable when performed in noisy environments. In this paper, a Hidden Markov Model (HMM) speech recognition system is used to assess the quality of speech signals recorded using an ear-mic combined with ANC. In addition, speech recognition itself can be of interest in many situations. For example, many functions in a vehicle can be controlled by instructions spoken by the user and speech controlled functionality improves safety. Hence, speech recognition is a growing field in vehicular technology. If the speech intelligibility and by that the speech recognition robustness could be increased in a speech recognition controlled system implemented in a vehicle, the safety would be further improved.

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• 263. Westerlund, Nils
Low Distortion SNR–Based Speech Enhancement Employing Critical Band Filter Banks2003Conference paper (Refereed)

This paper presents a speech enhancement method for personal communication where the input signal is divided into a number of subbands that are individually and adaptively weighted in time domain according to a short term SNR estimate in each subband at every time instant. Hence the name Adaptive Gain Equalizer. The method is focused on speech enhancement, acting as a speech booster, and remains idle when the SNR in a particular subband is low. Hence, background artifacts are eliminated and there is no need for a Voice Activity Detector (VAD). This paper also presents a method for least squares design of nonuniformfilter banks for application in subband signal processing. Design objectives aim to optimize the filter bank frequency response while minimizing subband and output aliasing. Critical band filter banks with increasing bandwidth are designed and evaluated with the speech enhancement method.

• 264. Winberg, Mathias
Speech Enhancement and Noise Combat1999Licentiate thesis, comprehensive summary (Other academic)
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• 265. Winberg, Mathias
Active Control of Engine Vibrations in a Collins Class Submarine2003Report (Other academic)

Submarine manufacturers invest considerable effort and money in improving the attenuation of noise radiated by submarines into the water. A number of different noise sources contribute to the total acoustic signature on an advanced submarine, such as the Collins Class. This report focuses on the problem associated with low order harmonics generated by the main diesel engines, transmitted through the engine mounts into the hull, and subsequently radiating into the water. The diesel engine mount consists of a two stage isolation system with a large intermediate mass, weighing approximately $500$ kg. Each engine is attached to the hull by eight mounts. The main intention of the work described here is to experimentally investigate the use of active vibration control to minimize the vibratory energy of the intermediate mass in all six degrees of freedom, thereby also minimizing the vibratory energy transmitted to the hull. This approach involves the use of seven inertial actuators, mounted on the intermediate mass, to produce the cancelling vibration field. A test rig, similar to one of the diesel engine mounts on the actual submarine, was constructed in the laboratory at The University of Adelaide and used for the real-time active control experiments described here. Several different set-ups were evaluated using primary vibrations with the same absolute levels and frequency content as measured on board a Collins class submarine. The controller used in the real-time experiments is based on the filtered X-LMS algorithm, which is a feed-forward algorithm working in the time domain.It is demonstrated here that active vibration control of the intermediate mass of the existing diesel engine mounts is a practical way of reducing the 1.0, 1.5 and 2.0 engine order components in the noise spectrum radiated as a result of vibration transmission through the engine mounts.

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• 266. Winberg, Mathias
A Feasibility Study of Applying Active Noise Control to a Fork-lift2003Conference paper (Refereed)

Modern fork-lift trucks incorporate a wide range of sophisticated passive techniques to reduce the noise inside the cabin. By the use of active noise control technologies it may be possible to even further reduce the noise level in total but also to attenuate certain tonal components, thereby improving the comfort for the truck driver. The paper discusses the noise situation inside the truck cabin as well as the active noise control strategy for a practical fork-lift truck installation. A feedforward control approach is used, along with several possible reference signals, forming a multiple reference controller.

• 267. Winberg, Mathias
An Active Headset Based on Hybrid Active Noise Control Technology2001Conference paper (Refereed)

In helicopters, the low frequency noise generated by the rotors and engines often masks and jeopardizes safe communication. In addition, pilots are likely to su?er from damage to their hearing due to the high sound levels in the headset produced to overcome the noise caused by increased speaker levels. A feasible approach is to reduce the low frequency noise by using active techniques combined with a method for reducing the noise in the intercom microphone signal, with lower speaker levels as a result. Helicopter noise consists of tonal components embedded in broadband noise. In order to achieve an e?cient attenuation of the primary noise inside the headset, a combination of a digital feedforward controller and an analog feedback controller is employed. Spectral Subtraction is used to suppress the background noise in speech signals. This paper evaluates a combination of the two techniques and their application to real data.

• 268. Winberg, Mathias
AVIIS, ACTIVE VIBRATION ISOLATION IN SHIPS; AN ASAC APPROACH1999Conference paper (Refereed)

Engine induced sound and vibration levels in boats for professional and leisure use is in many cases unacceptably high in term of comfort and the environment. Classical methods for passive treatment are normally less effective due to the low frequency content and often leads to an increase in weight. This contradicts the requirements for lower weight for increased speed. More efficient vibration damping methods must therefore be found. With active engine mounts, it is possible to achieve a decrease in the vibrations even when the hull is not very stiff. This is especially important in marine applications since the engines are mounted on weak and light structures. The AVIIS project aims at investigating the effects of a combined passive/active engine mount for use in boats. A Storebro 36 Royal Cruiser with two Volvo Penta engines has been used in the project. An ASAC (Active Structure Acoustic Control) approach has been used. The sound level in the cabin will be minimized by controlling the vibrations in the hull, in this way the active engine mounts only isolates the vibrations that couples good to the sound radiated. This paper will present the key results from the tests with the combined passive/active engine mounts on board the boat.

• 269. Winberg, Mathias
Active Control of Engine Induced Noise in a Naval Application2001Conference paper (Refereed)

Engine induced sound and vibration levels in boats for professional and leisure use is in many cases unacceptably high in term of comfort and the environment. Classical methods for passive treatment are normally less effective due to the low frequency content and often leads to an increase in weight. This contradicts the requirements for lower weight for increased speed. More efficient vibration damping methods must therefore be found. With active engine mounts, it is possible to achieve a decrease in the vibrations even when the hull is not very stiff. This is especially important in marine applications since the engines are mounted on weak and light structures. The AVIIS project aims at investigating the effects of a combined passive/active engine mount for use in boats. A Storebro 36 Royal Cruiser with two Volvo Penta engines has been used in the project. An ASAC (Active Structure Acoustic Control) approach has been used. The sound level in the cabin will be minimized by controlling the vibrations in the hull, in this way the active engine mounts only isolates the vibrations that couples good to the sound radiated. This paper will present the key results from the tests with the combined passive/active engine mounts on board the boat.

• 270. Winberg, Mathias
CONTROL APPROACHES FOR ACTIVE NOISE AND VIBRATION CONTROL IN A NAVAL APPLICATION2000Conference paper (Refereed)

Engine induced sound and vibration levels in commercial and leisure boats is in many cases unacceptably high in terms of passenger comfort and environmental noise. Classical methods of passive treatment are generally less effective due to the low frequency noise content, and often lead to an increase in structural weight. This than contradicts the low weight requirements for increased travel speed. More efficient vibration damping methods must therefore be found. With active engine mounts, it is possible to achieve a decrease in the vibrations even when the hull is not very stiff. This is especially important in marine applications since the engines are mounted on weak and light structures. The AVIIS (Active Vibration Isolation In Ships) project aims at investigating the effects of a combined passive/active engine mount for use in boats. A Storebro 36 Royal Cruiser with two Volvo Penta engines has been used in the project. This paper presents a comparison between a local vibration control at each engine mount contra a more global acoustic control, a ASAC (Active Structure Acoustic Control) approach. The advantage of acoustic control is that only the vibrations that couple strongly to the sound radiated will be attenuated. This also leads to a decrease in power consumption, which is important in a vessel. The calculations are based on real data from the Storebro boat.

• 271. Winberg, Mathias
A New Passive/Active Hybrid Headset for a Helicopter Application1999In: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 4, no 2Article in journal (Refereed)

In helicopters, the low frequency noise generated by the rotors and engines often masks and jeopardizes safe communication. In addition, pilots are likely to suffer from damage to their hearing due to the high sound levels in the headset produced to overcome the noise caused by increased speaker levels. A feasible approach is to reduce the low frequency noise by using active techniques combined with a method for reducing the noise in the intercom microphone signal, with lower speaker levels as a result. Helicopter noise consists of tonal components embedded in broadband noise. In order to achieve an efficient attenuation of the primary noise inside the headset, a combination of a digital feedforward controller and an analog feedback controller is employed. Spectral Subtraction is used to suppress the background noise in speech signals. This paper evaluates a combination of the two techniques and their application to real data.

• 272. Winberg, Mathias
Inertial Mass Active Mounts Used in a Marine Application1999Conference paper (Refereed)

Different actuators have been developed for active noise cancellation. For volumetric applications, a loudspeaker is usually used. When the noise is induced by engines it may be more efficient to work on the noise source itself, the engine vibration. In this case it is important to have an actuator that can counteract the vibration, not the sound field. In the research project AVIIS (Active Vibration Isolation In Ships), such an actuator has been developed. The actuator is an electrodynamic, inertial mass type shaker, designed and tuned for this project. The boat used in these experiments is a Storebro Royal Cruiser 33, powered by two Volvo Penta TAMD engines. Each engine is mounted to the hull in four points. Prior research, \cite{thomas1} shows that the main transmission paths for the vibrations from the engine and the propeller to the hull, are through these mounting points. Once the hull is excited, a lot of sound and annoying noise is produced in the cabin. The main idea is to isolate these vibrations from the hull by adding a combined active and passive engine mount that will control the vibrations by minimizing the sound field in the cabin using microphones as error sensors, a so called ASAC (Active Structural Acoustic Control) approach.

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• 273. Winberg, Mathias
Sound Measurements Using a Large Microphone Array1999Conference paper (Refereed)

Sound levels in closed cavities, such as on the command bridge of a military boat, are in many cases unacceptably high in terms of comfort and the sound environment. To be able to determine which means of sound reduction should be applied, it is important to have a good understanding of \ the origin of the sound field, and especially which parts in the cavity that have the highest sound levels. Today's hardware can handle large amounts of data. Therefore, a method using a large number of microphones is a good alternative for sound intensity measurements. Such a method has been used in this project. The method is evaluated on a Swedish minesweeper with a microphone matrix that consisted of 730 microphones positions, using 32 microphones at a time. The sound measurements are processed on a software platform via a dataacquisition system. This paper presents the background to the sound problem, the data collection and the analysis. Several sound plots will be presented on total sound levels and narrowband analysis, with conclusions on key sound sources etc. Conclusions on the experiences from this large measurement project will also be drawn.

• 274. Wu, Felix

This informational draft describes the DPAC (Data Packet Access Control) framework, potentially under PANA, to efficiently control "data packets" to access the network. Instead of using potentially more expensive crypto-based mechanisms such as IPSec (layer 3) or IEEE 802.11i (layer 2), DPAC introduces the possibility of using and negotiating a range of light-weight per-data-packet source authentication methods to control the data packets from PANA Clients (PaC). In DPAC, each data packet sent from PaCs to Enhanced Point (EP) can be classified, with high probability, as either valid or invalid. Furthermore, under this framework, it is possible for EP and PAA to account reliably on the network usage of each PaC.

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• 275. Yermeche, Zohra
A Calibrated Subband Beamforming Algorithm for Speech Enhancement2002Conference paper (Refereed)

This paper proposes a new calibrated adaptive frequency domain beamformer for speech enhancement. The beamformer is based on the principle of a soft constraint formed from calibration data, rather than precalculated from free-field assumptions. The benefit is that the real room acoustical properties will be taken into account. The proposed algorithm continuously estimates the spatial information for each frequency band, based on weighting of the received data. The update of the beamforming weights is done recursively where the initial precalculated correlation estimates of the speech constitute a soft constraint. The soft constraint secures the spatial-temporal passage of the desired source signal, without the need of any speech detection. The performance is evaluated in real world scenarios, both in a car-and a restaurant- environment. Interference and noise sup-pression of more than 15 dB is accomplished, while very small distortion is measured for the source signal.

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• 276. Zang, Zhuquan
Complex Domain Digital Filter Design with Max-Min Type Amplitude Constraints and Group Delay Specifications2001Conference paper (Refereed)

This paper considers the design of a digital filter with prescribed magnitude and group delay specifications. Our aim is to devise suitable methods/algorithms which are useful for the design of a set of filters for multiuser communication systems. To do this, we formulate our filter design problem as a constrained L2 space minimization problem in which the performance requirement on the group delay and magnitude in the passband are treated as constraints while minimizing the L2 norm of the error function between the designed and the desired filters. Methods for solving the proposed nonlinear and nonconvex optimization problem are outlined. Numerical results are presented to illustrate the usefulness of the proposed method.

• 277. Zang, Zhuquan
Design of Digital Filters with Amplitude and Group Delay Specifications2001Conference paper (Refereed)

This paper considers the design of a digital filter with prescribed magnitude and group delay specifications. First, we outline the derivation of the phase and group delay functions of an Nth order digital Laguerre filter and show that the group delay function of the filter can be written as a ratio of quadratic functions in the filter coefficients. Then, we formulate our filter design problem as a constrained L2 space minimization problem in which the performance requirement on the group delay and magnitude in the passband are treated as constraints while minimizing the L2 norm of the error function between the designed and the desired filters. Methods for solving the proposed nonlinear optimization problem axe outlined. Numerical results are presented to illustrate the usefulness of the proposed method. As a special case, corresponding results for general FIR filters axe also derived

• 278. Zhao, Fan
RBWA: an efficient random-bit window-based authentication protocol2003Conference paper (Refereed)

Given the wide and rapid depolyment of "visitor networks", how to authenticate the user and account the usage on the per-packet basis securely and yet efficiently is still a challenging problem. In this paper, we explore the tradoff between performance and security, and propose a per-data-packet authentication and access control called RBWA (Random Bit Window based Authentication). Deployed in the IP layer, RBWA can work with various underlying link layer specific mechanisms and network topologies. And comparing to IPSec, it dramatically reduces the overhead and power consumption by adding only a few bits to each packet. Furthermore, RBWA is strong against a suite of attacks such as replay attack, Denial-of-Service attack and spoofing etc. In particular, a robust anti-replay window scheme is developed to counter the svere packet reordering. The performance of RBWA is evaluated via the simulation.

• 279.
Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
Identification and Analysis of Nonlinear Systems2003Independent thesis Advanced level (degree of Master (One Year))Student thesis

In classical mechanical engineering the predominant group of system analysis and identification tools relies on Linear Systems, where research have been carried out for over half a century. Usage of Linear Systems is most widely spread, often due to its simple mathematics and formulation for many engineering problems. Although linearizing is a means for simplifying a problem, it will introduce more or less severe modelling errors. In some cases the errors due to linearizing are too large to be practically acceptable, and therefore nonlinear structures and models are sometimes introduced. This thesis aims in implementing and evaluating some popular methods and algorithms for nonlinear structure analysis and identification, with emphasis on systems having nonlinear terms. Preferably the algorithms should be optimized in their computational load. The result are several algorithms for nonlinear analysis and identification. The ones giving best results were the frequency based methods Reverse Path and a Frequency Domain Structure Selection Method (FDSSA). The time domain based method, Nonlinear Autoregressive Moving Average with Exogenous Input (NARMAX), in which a lot of hope had been put, did perform very well in giving good system descriptions, but due to its nonphysical representation it was not suitable for usage in this thesis. The algorithms and methods were finally applied for two cases, a four system black-box case and an experimental test-rig case. The methods did perform well in three out of four systems in the first case, but the methods did not perform well for the second case, due to problems in applying correct levels of excitation force at the test-rig’s resonance frequencies.

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