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  • 451. Yatawara, Yeshan
    et al.
    Caldera, Manora
    Kusuma, Tubagus Maulana
    Zepernick, Hans-Jürgen
    Unequal Error Protection for ROI Coded Images over Fading Channels2005Conference paper (Refereed)
    Abstract [en]

    Region of interest (ROI) coding is a feature supported by the Joint Photographic Experts Group 2000 (JPEG2000) image compression standard and allows particular regions of interest within an image to be compressed at a higher quality than the rest of the image. In this paper, unequal error protection (UEP) is proposed for ROI coded JPEG2000 images as a technique for providing increased resilience against the effects of transmission errors over a wireless communications channel. The hierarchical nature of an ROI coded JPEG2000 code-stream lends itself to the use of UEP whereby the important bits of the code-stream are protected with a strong code while the less important bits are protected with a weaker code. Simulation results obtained using symbol-by-symbol maximum a posteriori probability (MAP) decoding demonstrate that the use of UEP offers significant gains in terms of the peak signal to noise ratio (PSNR) and the percentage of readable files. Moreover, the use of ROI-based UEP leads to reduced computational complexity at the receiver.

  • 452. Yermeche, Zohra
    Soft-Constrained Subband Beamforming for Speech Enhancement2007Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    New speech acquisition applications are emerging as a result of advances in technology and the prevalence of mobile communication. While today voice control of consumer equipment is becoming a reality, communication technology has extended voice connectivity to personal computers and mobile communication devices with the aim of enabling natural communication in a variety of environments such as cars, restaurants and offices. The comfort and flexibility provided through the hands-free acquisition of speech in mobile telephony, speech recognition and hearing aids require robust techniques to deal with problems of environmental noise, reverberation, acoustic feedback and other interfering sounds which corrupt the received speech. For mobile environments, speech enhancement techniques should also provide an adaptation capacity to speaker motion with no perceptible degradations of the original speech. In this thesis, multi-microphone techniques for speech enhancement are developed. First, a framework for constrained beamforming is introduced. This framework allows us to control the tradeoff relationship between noise reduction, dereverberation and speech degradation. A constraint on the power minimization of the beamformer’s output is formulated to guarantee the integrity of the desired signal. It is shown that the robustness towards microphone mismatch of the soft-constrained beamforming structure is guaranteed by modeling the source as spatially spread. A subband recursive least-squares (RLS) beamformer is investigated and evaluated in real handsfree acoustical environments. The proposed methodology is defined to perform background noise and interference reduction, while a soft constraint built from calibration data in low noise conditions guarantees the undistorted filtering of the desired signal. This adaptive structure allows for a tracking of the noise characteristics, so as to efficiently accomplish its attenuation. A subband beamforming structure is used to improve the performance of the system and reduce the computational complexity. A real-time DSP implementation is described and evaluated for dual microphone speech enhancement. Furthermore, a novel blind soft-constrained beamforming approach for moving source speech enhancement is presented. It is based on a soft constraint defined for a delay-spread corresponding to a volume around the speech source location. A new speech-oriented time-delay estimation algorithm is combined with the beamformer to allow for speaker movement. The proposed method does not require any calibration data, knowledge of the array manifold or any other characteristics of the acoustical environment. Hence, it provides means to blindly enhance a dominant speaker in adverse noise conditions. This structure is further developed to allow for the detection and enhancement of multiple dominant speakers in a mixture of interferences and background noise. The use of a frequency-dependent constraint region opens the path for a trade-off between noise suppression and speech integrity.

  • 453. Yermeche, Zohra
    Subband Beamforming for Speech Enhancement in Hands-Free Communication2004Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Speech enhancement by means of microphone array signal processing has a major role in voice communication applications such as audio-conferencing, hands-free telephony, voice recognition and hearing aids. In these communication scenarios, the speaker is positioned at a remote distance from the microphones, which causes problems of environment noise and interfering sound corrupting the received speech. Additionally, reverberations of the voice from walls or ceilings, also impairs the received speech signal. In the case of a duplex communication, the acoustic feedback constitutes another disturbance for the talker who hears his or her voice echoed. Successful speech enhancement solutions should achieve speech dereverberation, efficient noise and interference reduction, and for mobile environments, they should also provide an adaptation capacity to speaker motion. Microphone arrays spatially sample the sound pressure field. When combined with spatio-temporal filtering techniques known as {\em beamforming}, they can extract the sound source information from signals, of which only a mixture is observed. This is based on the inherent ability of sensor arrays to exploit the spatial correlation of multiple received signals. A subband beamforming structure can be used in order to improve the performance of the time-domain filters and reduce their computational complexity. Each of the received signals is decomposed into a set of narrow-band signals and the filtering operations of the beamformer are performed for each frequency band separately. The output of the subband beamformers are then used to reconstruct a full-band output signal. In this thesis an adaptive subband RLS beamforming approach is investigated and evaluated in real hands-free acoustical environments. The proposed methodology is defined such to perform background noise and acoustic coupling reduction, while producing an undistorted filtered version of the signal originating from a desired location. The beamformer recursively minimizes a Least Squares error based on the continuously received data. This adaptive structure allows for a tracking of the noise characteristics, such to accomplish its attenuation in an efficient manner. A soft constraint built from calibration data in low noise conditions guarantee the integrity of the desired signal without the need of any speech detection. Additionally, a new spatial filter bank design method for beamforming applications, which includes the constraint of signal passage at one position and closing in other undesired positions, is suggested. Furthermore, to allow for source mobility tracking, a soft constrained beamforming approach with built-in speaker localization, is proposed. The source of interest is modelled as a cluster of point sources and source motion is accommodated by revising the point source cluster. Real speech signals are used in the simulations and results show accurate speaker movement tractability with maintained noise and interference suppression of about 10-15 dB, when using a four-microphone array.

  • 454. Yermeche, Zohra
    et al.
    Cornelius, Per
    Grbic, Nedelko
    Claesson, Ingvar
    Spatial Filter Bank Design for Speech Enhancement Beamforming Applications2004Conference paper (Refereed)
    Abstract [en]

    In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constraint of signal passage at one position (and closing in other positions corresponding to known disturbing sources). By performing the directional opening towards the desired location in the fixed filter bank structure, the beamformer is left with the task of tracking and suppressing the continuously emerging noise sources. This algorithm has been implemented in MATLAB and tested on real speech recordings conducted in a car hands-free communication situation. Results show that a reduction of the total complexity can be achieved while maintaining the noise suppression performance and reducing the speech distortion.

  • 455. Yermeche, Zohra
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Bind Subband Beamforming with Time-delay Constraints for Moving Source Speech Enhancement2007In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 15, no 8, p. 2360-2372 Article in journal (Refereed)
    Abstract [en]

    A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.

  • 456. Yermeche, Zohra
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Blind Subband Beamforming for Speech Enhancement of Multiple Speakers2006Conference paper (Refereed)
    Abstract [en]

    A new blind microphone array method to enhance speech signals generated by multiple sources in a noisy environment is proposed. This approach is based on a two-stage scheme. A subband time-delay estimation algorithm is first used to localize the dominant speech sources. The speech enhancement is performed in a second stage, based on the acquired spatial information, by means of a spatially constrained subband beamformer. The robustness of this structure is ensured by the spatial constraint constructed to include the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Such scheme also allows for an efficient adaptation of the beamformer to speakers movement. The proposed subband approach for time-delay estimation exploits the sparseness of speech signals in the time-frequency domain to localize multiple speakers simultaneously. It also provides means to select the number of target sources. Evaluation in a real environment shows promising results.

  • 457. Yermeche, Zohra
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Blind Subband Beamforming With Time-Delay Constraints for Moving Source Speech Enhancement2007In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 15, no 8, p. 2360-2372Article in journal (Refereed)
    Abstract [en]

    A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.

  • 458. Yermeche, Zohra
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Moving Source Speech Enhancement Using Time-Delay Estimation2005Conference paper (Refereed)
    Abstract [en]

    This paper presents a new constrained subband beamforming algorithm to enhance speech signals generated by a moving source in a noisy environment. The beamformer is based on the principle of a soft constraint calculated from an estimated source position. The soft constraint secures the spatial-temporal passage of the desired source signal in the adaptive update of the beamforming weights and guaranties the full rank property of the covariance matrix inverted in the update. This approach allows for an efficient adaptation of the beamformer to speaker movement by using a tracking algorithm for sound source time-delay estimation. The proposed method has the benefit of taking into consideration the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Evaluation in a real environment with a moving speaker in a hands-free situation shows up to 10~dB noise suppression and 20~dB interference suppression within the conventional telephone bandwidth. This is achieved with a negligible impact on target signal distortion.

  • 459. Yermeche, Zohra
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Speech enhancement of multiple moving sources based on subband clustering time-delay estimation2006Conference paper (Refereed)
    Abstract [en]

    A new robust blind microphone array method to enhance speech signals generated by multiple moving sources in a noisy environment is presented. This approach is based on a two-stage scheme. A subband clustering time-delay estimation algorithm is first used to localize the dominant speech sources. The speech enhancement is performed in a second stage, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The robustness of this structure is ensured by the spatial constraint constructed to include the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Such scheme also allows for an efficient adaptation of the beamformer to speakers movement. The proposed subband clustering approach for time-delay estimation exploits the sparseness of speech signals in the time-frequency domain to localize multiple speakers simultaneously. It also provides means to select the number of target sources. Evaluation in a real environment with moving speakers shows promising results.

  • 460. Yiu, K. F. C
    et al.
    Chan, K. Y
    Grbic, Nedelko
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Nordholm, Sven
    A Hybrid Design Of Beamformers For Voice Control Devices2012In: Pacific Journal of Optimization, ISSN 1348-9151, E-ISSN 1349-8169, Vol. 8, no 3, p. 533-544Article in journal (Refereed)
    Abstract [en]

    In this paper, a new approach to designing beamformers for voice control device is proposed. It is well-known that under a strong near-field noise with low signal-to-noise ratios (SNR), the performance of speech recognition is deteriorated significantly. However, designing the beamformer for enhancing speech recognition is a slow process and might not be adapted easily to the changing noise environment. In order to lower the complexity of the design, we intend to exploit the combination of existing optimal beamformer designs, which can be implemented in parallel. These include the least-squares technique and the signal-to-noise ratio maximization technique. We show here that for a given pre-trained speech recognizer and for a finite set of speech commands, neither method has a satisfactory performance in speech recognition accuracy under very low SNRs. However, since the two techniques have different characteristics in speech distortion and noise suppression, we show that it is possible to enhance the speech recognition accuracy by combining these two optimal designs.

  • 461. Yiu, Ka Fai Cedric
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Teo, Kok Ley
    A hybrid method for the design of oversampled uniform DFT filter banks2006In: Signal Processing, ISSN 0165-1684 , Vol. 86, no 7, p. 1355-1364Article in journal (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to speed up the convergence and to lower the computational complexity of time domain adaptive filters. However, subband processing causes signal degradations due to aliasing effects and amplitude distortions. This problem is unavoidable due to further filtering operations in subbands. In this paper, the problem of aliasing effect and amplitude distortion is studied. The prototype filter design problem is formulated as a multi-criteria optimization problem and all the Pareto optima are sought. Since the problem is highly nonlinear and nonsmooth, a new hybrid optimization method is proposed. Different prototype filters are used and their performances are compared. Moreover, the effect of the number of subbands, the oversampling factors and the length of prototype filter are also studied. We find that prototype filters designed via Kaiser or Dolph–Chebyshev window provide the best overall performance. Also, there is a critical oversampling factor beyond which the improvement in performance is not justified. Finally, if the length of the prototype filter increases with the number of subbands, an increase in the subband level will not deteriorate the performance.

  • 462. Yiu, Ka Fai Cedric
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Teo, Kok Ley
    Multicriteria Design of Oversampled Uniform DFT Filter Banks2004In: IEEE Signal Processing Letters, ISSN 1070-9908, E-ISSN 1558-2361, Vol. 11, no 6, p. 541-544Article in journal (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. However, subband processing causes signal degradations due to aliasing effects and amplitude distortions. This problem is unavoidable due to further filtering operations in subbands. In this letter, the problems of aliasing effect and amplitude distortion are studied. Prototype filters which are optimized with respect to those properties are designed and their performances are compared. Moreover, the effect of the number of subbands, the oversampling factors and the length of the prototype filter are also studied. Using the multicriteria formulation, all Pareto optimums are sought via the nonlinear programming technique. We find that the prototype filter designed via the Kaiser window provides the best overall performance among the methods we studied. Also, there is a critical oversampling factor beyond which the improvement of performance is diminishing. Finally, if the length of the prototype filter increases with the number of subbands, an increase in the number of subbands will not deteriorate the performance.

  • 463. Yiu, Ka Fai Cedric
    et al.
    Ho, Chun Hok
    Grbic, Nedelko
    Lu, Yao
    Shi, Xiaoxiang
    Luk, Wayne
    Reconfigurable Acceleration of Microphone Array Algorithms for Speech Enhancement2008Conference paper (Refereed)
    Abstract [en]

    Microphone arrays play an important role in noise reduction and speech enhancement. Their algorithms are based on beamforming, which reduces the level of localized and ambient noise signals while minimizing distortion to speech from the desired direction via spatial filtering. This paper describes a class of subband beamforming algorithms. The similarity between different algorithms is discussed. To enhance computational efficiency, the algorithms are implemented in frequency domain. A hardware architecture, with bitwidth optimization, is proposed to support the algorithms. An implementation with 7 instances on a Xilinx XC4VSX55 FPGA at 175MHz can run 41.7 times faster than the corresponding pure software implementation on a 3.2GHz Pentium 4 PC. ©2008 IEEE.

  • 464. Yiu, KFC
    et al.
    Gao, MJ
    Shiu, TJ
    Wu, SY
    Tran, Trung Quang
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Claesson, Ingvar
    Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering.
    A fast algorithm for the optimal design of high accuracy windows in signal processing2013In: Optimization Methods and Software, ISSN 1055-6788, E-ISSN 1029-4937, Vol. 28, no 4, p. 900-916Article in journal (Refereed)
    Abstract [en]

    This paper presents a new optimal window design method with a general window design specification for the passband and stopband. The design problem is formulated as a semi-infinite linear programming problem. With suitable discretizations, an exchange algorithm is employed. The convergence of the proposed algorithm is established. In the formulation, since the stopband is minimized, the method can be employed for the design of very highly optimized windows.

  • 465.
    Yves, Hawa
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    A Study of Fast Backprojection Algorithm for UWB SAR and a Comparison between Fast- and Global Backprojection2008Independent thesis Advanced level (degree of Master (Two Years))Student thesis
    Abstract [en]

    In this thesis, we present a study of the Fast Backprojection (FBP) known to be a fast time-domain algorithm for image retrieval in Synthetic Aperture Radar (SAR). As a time domain algorithm, FBP possesses inherent advantages such as perfect motion compensation, unlimited scene size, wide bandwidth and ability to handle long integration angles. Although FBP reproduces SAR images on pixel-by-pixel basis, the processing time for FBP is reduced significant compared to the Global Backprojection (GBP) with a heavy computational load. For GBP, the number of operations to process a N×N SAR image with N aperture positions is proportional to N . Whereas, the number of operation required by FBP in the same case is reduced by a factor of sqrt(N) i.e. proportional to N2sqrt(N). We give a detailed explanation on how to implement FBP in Matlab in order to retrieve a SAR image. We explain about different techniques of interpolation used to obtain a high quality image. We also present a method to compare between interpolation techniques in term of SAR image quality and processing time. A comparison between FBP and GBP is also a topic to be discussed in this thesis. In this comparison, the processing time in theory and reality will be focused.

  • 466.
    Zepernick, Hans-Jürgen
    et al.
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Fiedler, Markus
    Blekinge Institute of Technology, School of Engineering, Department of Telecommunication Systems.
    Lundberg, Lars
    Blekinge Institute of Technology, School of Engineering, Department of Systems and Software Engineering.
    Pettersson, Mats
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Quality of Experience Based Cross-Layer Design of Mobile Video Systems2008Conference paper (Refereed)
    Abstract [en]

    This paper introduces and discusses quality of experience based cross-layer design of mobile video systems as a means of providing technologies for jointly analyzing, adopting, and optimizing system quality. The many benefits of our novel approaches over traditional concepts will range from efficient video processing techniques over advanced real-time scheduling algorithms, to networking and service level management techniques. This will lead to better service quality and resource utilization in mobile video systems.

  • 467. Zepernick, Hans-Jürgen
    et al.
    Finger, Adolf
    Pseudo Random Signal Processing: Theory and Application2005Book (Other academic)
  • 468. Zepernick, Hans-Jürgen
    et al.
    Finger, Adolf
    Pseudo Random Signal Processing: Theory and Application (Chinese translation edition)2007Book (Other academic)
    Abstract [en]

    Chinese translation edition of the book ''Pseudo Random Signal Processing: Theory and Application'' by Hans -Jürgen Zepernick and Adolf Finger, John Wiley & Sons, 2005.

  • 469.
    Zetterberg, Viktoria
    Blekinge Institute of Technology, School of Engineering, Department of Signal Processing.
    Passive Acoustic Monitoring of Moving Targets by Sparse Underwater Sensor Network2008Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Surveillance of sound activities in the sea is of great importance for many applications. A common way to monitor the sea is to register sound with the help of a group of closely placed sensors that form a one-dimensional or two-dimensional array. This is a practical and easy way to monitor sound in large and deep waters where sound spreads in nearly straight lines. In shallow waters, propagation patterns are more complex and also more difficult to predict. Baltic Sea shallow waters also involve large archipelagoes, which make monitoring more difficult. New possibilities to synchronize sensors with the aid of the global positioning system (GPS) facilitate the creation of larger sensor systems. By placing sensors at different positions over the entire water volume under observation, a minimization of the effects from complicated sound spreading is allowed. Near field processing can then also be applied since the sound source is registered by means of several sensors in positions close to the target. These large arrays will have three-dimensional capabilities and they will also provide a much higher resolution than traditional systems. In this thesis it is shown that signals registered hundreds of meters away from each other can be used to position moving sound within the sensor area. Measurement results are presented from four experiments in shallow waters 20 meters deep, located in the Baltic Sea. The measurements were performed with different setups, at different locations and during different seasons. The thesis comprises four parts. The first part presents a comparative study of two methods for estimating the time delay between two measurement signals, where the sound consists of noise and dominant tones. Effects from different geometric setups on the position estimate are shown. The result indicates that larger baselines give better positioning capabilities despite a larger uncertainty in the time delay estimation. The second part of this thesis concerns imaging, which means the projection of multiple pairwise correlation results into a two-dimensional plane. Results from a target placed at a distance equal to the length of the sensor baselines are shown. Weights that can be used to give different sensor pairs different importance in the summation image are investigated. In this case, the target was assumed to consist of one sound source. The third part of the thesis also concerns imaging, but presents results from a target placed at a distance smaller than the length of the sensor baselines. In this part, simulation shows that it is possible to position an object in a three-dimensional space using sensors placed in a two-dimensional span. The fourth and final part of this thesis also focuses on imaging and here the results indicate that several sound sources can be resolved for a target.

  • 470. Zetterberg, Viktoria
    et al.
    Pettersson, Mats
    Claesson, Ingvar
    Acoustic passive 3D imaging method for scattered array2006Conference paper (Refereed)
    Abstract [en]

    An acoustic passive positioning/image method using widely distributed sensors in a network is shown. Previous studies have shown that, despite decreased coherence (due to the larger sensor separation) between sensor signals, a scattered array can give a better positioning capabillity. The positioning method divides the water into volumes and uses a weighting of each sensor pair’s correlation (Generalized Cross Correlation) result to test each volume individually. Various aspects can govern the weight assigned to a sensor pair and we have in this paper used distance from sensor to target. By projecting the correlation results over the map an intersection point is found. By looking at different depths the intersection point is found at the layer where the sound source is located. 3D positioning though requires an end fire sensor pair close to the sound source. The more pairs of that kind, the better the position in 3D will be. The developed method is tested on both simulations and real measurement data from a measurement campaign along the Swedish coastline. In order to image moving targets by scattered sensors it is also important to Doppler compensate.

  • 471. Zetterberg, Viktoria
    et al.
    Pettersson, Mats
    Claesson, Ingvar
    Comparison between whitened generalized cross correlation and adaptive filter for time delay estimation with scattered arrays for passive positioning of moving targets in Baltic Sea shallow waters2005Conference paper (Refereed)
    Abstract [en]

    Results from an acoustic measurement campaign in shallow waters (at 22 meters depth) are given. Eight hydrophones, placed at the seabed, were used simultaneously to investigate how different distances between hydrophones affect the Time Delay Estimation (TDE). In order to position targets two TDE methods were applied. The used methods were the Generalized Cross Correlation (GCC) function weighted by the Phase Transform (PHAT) and an adaptive Normalized Least Mean Square (NLMS) filter estimating the TDE between pairs of sensors. The two methods gave approximately the same result, except for the largest baseline where the GCC PHAT performed better. In the study we found that increasing separation distance decrease the bandwidth of coherence between the signals, however the study also indicated that scattered arrays with large baselines has better positioning capability than those with small baselines.

  • 472. Zetterberg, Viktoria
    et al.
    Pettersson, Mats
    Tegberg, Lars
    Claesson, Ingvar
    Passive scattered array positioning method for underwater acoustic source2006Conference paper (Refereed)
    Abstract [en]

    This paper considers large scattered arrays for 3D underwater acoustic positioning and imaging, where the hydrophone sensors, in the array, form a network over the total area of interest. A method where the Generalized Cross Correlation (GCC) function is projected over the volume is being used, and 3D images are formed. In this paper we determine the discretization of these images so they fulfill the Nyqvist criteria. In order to detect and position targets by scattered sensors it is important to Doppler compensate and weight the different sensor pair results. In this paper we divide the water into volumes where Doppler compensation and pair weights are implemented for each volume individually. Various aspects can govern the weights assigned to a sensor pair and in this paper we have used opening angle between sensor pair and target. The baseline of the sensor pairs varies from 17 – 181 meters, therefore different frequency bandwidths for each sensor pair is used. The developed method is tested on both simulations and real measurement for a surface target.

  • 473. Åkesson, Henrik
    Active control of vibration and analysis of dynamic properties concerning machine tools2007Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Vibration in internal turning is a problem in the manufacturing industry. Vibrations appear under the excitation applied by the material deformation process during the machining of a workpiece. In order for a lathe to perform an internal turning or boring operation, for example, in a pre-drilled hole in a workpiece, it is generally required that the boring bar should be long and slender; therefore extra sensitive to vibrations. These vibrations will affect the result of machining, in particular the surface finish, also the tool life may be reduced. As a result of tool vibration, severe acoustic noise frequently occurs in the working environment. This thesis comprises three parts and the first part presents a method for active control of boring bar vibration. This method consists of an active boring bar controlled by, for example, an analog controller. The focus lies on the analog controller and the advantages that may be obtained from working in the analog domain. The controller is a lead-lag compensator with digitally controlled parameters, such as gain and phase. However, signals remain in the analog domain. In addition, the analog controller is compared with a digital adaptive controller and it is found that both controllers yield an attenuation of the vibration by up to 50 dB. The second part of this thesis concerns the dynamic properties of a clamped boring bar used by the industry. In order to design a robust controller for a certain system, knowledge about the system's dynamic properties is required. On the workshop floor, a boring bar is dismounted and remounted, and reconfiguration of boring bars will alter the dynamic properties of the clamped boring bar. The dynamic properties of a standard boring bar and an active boring bar for a number of possible clamping conditions, as well as for a linearized clamping have been investigated based on an experimental approach. Also simple Euler-Bernoulli modeling of clamped boring bars incorporating simple non-rigid models of the boring bar clamping are investigated. Initial simulations of nonlinear SDOF systems have been carried out: one with a signed squared stiffness and one with a cubic stiffness. The purpose of these simulations was to identify a nonlinearity that introduces a similar behavior in the SDOF system dynamics as the nonlinear behavior observed in the dynamic properties of a clamped boring bar. The third and final part of this thesis focuses on vibration analysis methods in engineering education. A signal analyzer (which is a commonly used instrument in signal processing and vibration analysis) was made accessible via the Internet. Assignments were developed for students to learn and practice vibration analysis on real signals from a real setup of a relevant structure; a clamped boring bar. Whilst the experimental setup was fixed, the instrument and sensor configuration nonetheless enable a variety of experiment, for example: excitation signal analysis, spectrum analysis and experimental modal analysis.

  • 474. Åkesson, Henrik
    Investigation of the Dynamic Properties of a Milling Structure; Using a Tool Holder with Moderate Overhang2009Conference paper (Refereed)
    Abstract [en]

    Vibration problem during metal cutting occurs frequently in the manufacturing industry. The vibration level depends on many different parameters such as material type, dimensions of the workpiece, rigidity of tooling structure, cutting data and operation mode. In milling the cutting process subject the tool to vibrations and having a milling tool holder with long overhang, will most likely results in high vibration levels. These vibrations have a consequence of reduced tool life, poor surface finishing and disturbing sound. An investigation of dynamic properties of a milling tool holder with moderate overhang has been carried out by means of experimental modal analysis and vibration analysis during operating mode. Both angular vibrations of the rotating tool and vibrations of the machine tool structure were examined during milling. Also, bending motion of the rotating milling tool holder was examined during cutting. This paper focuses on identifying the source/sources of the dominating milling vibration components and to identify that of these vibrations which may be related to the structural dynamic properties of the milling tool holder.

  • 475. Åkesson, Henrik
    et al.
    Brandt, Anders
    Håkansson, Lars
    Lagö, Thomas L
    Claesson, Ingvar
    Operational Modal Analysis of a Boring Bar During Cutting2005Conference paper (Other academic)
    Abstract [en]

    Internal turning or boring operation is a common metal working process that is usually associated with vibration problems. Vibration problems in internal turning considerably influence important factors such as surface quality, productivity, production costs, etc. In this paper we report analysis results from an operational modal analysis of a boring bar during cutting operation. The results are compared with operating deflection shape results and traditional modal analysis results of the same boring bar. Results show that the first bending modes in the directions of cutting speed and cutting depth are active in the operating vibrations. The three analysis methods result in similar conclusions.

  • 476. Åkesson, Henrik
    et al.
    Gustavsson, Ingvar
    Håkansson, Lars
    Claesson, Ingvar
    Remote Experimental Vibration Analysis of Mechanical Structures over the Internet2005Conference paper (Refereed)
    Abstract [en]

    Experimental vibration analysis is of significant importance for e.g. the automobile and aircraft industry. It provides detailed information concerning the actual dynamic properties of vibration, structures, etc. Commonly, information from experimental vibration analysis is used in the development or modification of structures, processes, etc. to obtain for instance a required dynamic behaviour. It usually involves analysis methods such as; spectrum analysis, correlation analysis, experimental modal analysis and operating deflection shapes analysis (ODS). Large experience is generally required to obtain reliable results with these methods. The best way to acquire such experience is to spend many hours in a laboratory supervised by an expert teacher. However, in engineering education the experimental resources are limited and the traditional way of conducting vibration experiments is to participate in time limited scheduled lab sessions. Internet, however, provides the opportunity for engineering students to access the practical and theoretical knowledge advancement in experimental vibration analysis that is highly attractive for the industry. Laboratory exercises in, for example, experimental vibration analysis and signal processing courses, can now be performed remotely using real equipment. Advanced vibration experiments have been conducted over the Internet at Blekinge Institute of Technology, Sweden; the experiments have been carried out using experimental hardware located in a small closed laboratory. Exercises are adapted to on-campus students as well as distance learning engineers in continuing education programs. A new possibility to directly integrate vibration experiments into lectures given by expert teachers appears and after each lecture the students can repeat and elaborate on the experiments. Thus, enabling the students to carry out the experiments within a course at home using the time they require for sufficient comprehension. In this paper the remote experimental vibration analysis laboratory and its possibilities will be presented.

  • 477. Åkesson, Henrik
    et al.
    Håkansson, Lars
    Gustavsson, Ingvar
    Zackrisson, Johan
    Claesson, Ingvar
    Lagö, Thomas L
    Vibration Analysis of Mechanical Structures over the Internet Integrated into Engineering Education2006Conference paper (Refereed)
    Abstract [en]

    Experimental vibration analysis is one of the most important tools for analyzing dynamic properties of mechanical structures. The information from experimental vibration analysis is used in the development of products to obtain a required dynamic behavior, or for instance classify vibration problems in different public, industrial environments etc. In order to carry out such experiments with high quality, knowledge about different analysis methods is of great importance. Also a large experimental experience is required to obtain reliable results as in any field. In engineering education experiment using real mechanical structures and equipment is of significance for the learning process. In traditional university laboratories students conduct experiments under the supervision of an instructor. However, a trend towards decreasing investments to maintain these laboratories can be seen all over the world and instead an increase of simulations or theoretical experiments is replacing these hands on experiments due to cost. Blekinge Institute of Technology (BTH), Sweden provide the opportunity for engineering students to remotely access the practical and theoretical knowledge advancement in experimental vibration analysis that is highly attractive for the industry. Remote laboratory exercises are to day a reality at BTH and are a complement to on-campus laboratory experiments, increasing the availability of the instruments. Previously vibration experiments have been conducted over the Internet at BTH, using experimental hardware located in a small closed laboratory. A further step towards a more user-friendly interface has been developed, increasing the feeling of being in front of a real instrument instead of a virtual front panel. This paper presents a new remote vibration laboratory and how remote experimental vibration analysis has been integrated in the engineering education as a complement to ordinary lessons and experiments in traditional laboratories.

  • 478. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Claesson, Ingvar
    Håkansson, Lars
    On the Development of a Simple and Robust Active Control System for Boring Bar Vibration in Industry2007In: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 12, no 4, p. 139-152Article in journal (Refereed)
    Abstract [en]

    Vibration in internal turning is a problem in the manufacturing industry. A digital adaptive controller for the active control of boring bar vibration may not be a suficient solution to the problem. The inherent delay in a digital adaptive controller delays control authority and may result in tool failure when the load applied by the workpiece on the tool changes abruptly, e.g. in the engagement phase of the cutting edge. A robust analog controller, based on a lead-lag compensator, with simple adjustable gain and phase, suitable for the industry application, has been developed. Also, the basic principle of an active boring bar with embedded actuator is addressed. The performance and robustness of the developed controller has been investigated and compared with an adaptive digital controller based on the feedback filtered-x algorithm. In addition, this paper takes into account those variations in boring bar dynamics which are likely to occur in industry; for example, when the boring bars is clamped in a lathe. Both the analog and the digital controller manage to reduce the boring bar vibration level by up to approximately 50 dB.

  • 479. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Analog and Digital Approaches of Attenuation Boring Bar Vibrations during Metal Cutting Operations2005Conference paper (Refereed)
    Abstract [en]

    In metal cutting operations tool vibration is a frequent problem. Vibrations appear under the excitation applied by the material deformation process during the machining of a workpiece. In order to perform an internal turning or boring operation, e.g. in a pre-drilled hole in a workpiece, it is generally required that the boring bar should be long and slender; therefore it is easily subjected to vibrations. These vibrations will affect the result of machining, in particular the surface finish, and also the tool life may be reduced. As a result of tool vibration severe acoustic noise frequently occurs in the working environment. The vibration problem is to a large extent related to the boring bar low-order fundamental bending modes. In order to control boring bar vibrations in the primary cutting direction an analog and a digital feedback controllers have been used. In both approaches an active boring bar with a built-in actuator has been used. In order to measure the response of the active boring bar, accelerometers are mounted nearby the area, where the excitation force is applied due to material deformation process. This means that in either approach the boring bar vibrations are attenuated actively by the controlled secondary anti-vibrations induced by the actuator. The digital controller, based on the feedback filtered-x LMS-algorithm, manages to reduce the boring bar vibrations in the primary cutting direction by up to approximately 44 dB at the first bending resonance frequency. The analog controller, based on a flexible orthogonal gain and phase lag compensation, results in an attenuation of the boring bar vibrations in the primary cutting direction by up to approximately 40 dB at the first bending resonance frequency. Both controllers also suppress all the harmonics of the first bending resonance frequency. Vibration attenuation performances of the two controllers are compared and discussed.

  • 480. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Advantages and Drawbacks Using Different Sensors in Feedback Control in Active Boring Bar Applications2008Conference paper (Refereed)
    Abstract [en]

    Several prototypes of an active boring bar for industrial applications have been produced for the reduction of cumbersome vibration problems in metal cutting processes. The active boring bar is used in lathe application to machine deep and precise holes in workpieces where long and slender boring bars are required. A vibration sensor embedded in the boring bar together with an embedded actuator is utilized in order to enable active feedback control. This paper discusses different techniques of measuring boring bar vibrations and their influence on the feedback control scheme as well as on the active control performance. Also the dynamic properties of the boring bar are considered when utilizing the different sensor techniques. The sensors that have been used are; accelerometer, strain gauges and piezo film.

  • 481. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Analog versus Digital Control of Boring Bar Vibration2005Conference paper (Refereed)
    Abstract [en]

    In workshops where metal cutting is performed, the machining processes frequently introduces productivity degrading vibration problems and annoying sound, sometimes almost at unbearable levels. Besides producing disturbing noise, the vibrations affect the surface finish of the workpiece and the tool life. Two different approaches based on feedback control are investigated, both applied for the control of an active boring bar. The first approach is based on a digital adaptive feedback controller; the feedback filtered-X LMS algorithm. The second approach is based on an analog controller; a feedback controller with gain and phase orthogonally adjustable, thus flexible for the control of systems with different dynamic properties. Based on open loop frequency response function estimates, robustness and stability of both the digital and the analog feedback control system are discussed. A comparison of the two controllers concerning their boring bar vibration attenuation performance shows that the analog controller attenuates the vibration in same order of magnitude as the digital controller which is approximately by up to 40 dB.

  • 482. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Comparison between different controllers when having abrupt changes in a process2006Conference paper (Refereed)
    Abstract [en]

    When metal cutting is performed, the machining processes introduce productivity degrading vibration. By applying an active control scheme, theses vibrations can be reduced significantly with the result of improved surface finish of the work piece and increased tool life. Due to the large difference of boundary conditions during the machining process, a controller fast enough to follow these changes is needed, for example from no cutting to cutting - the actual engagement of the cutting process. If the controller does not success to follow fast enough, the tool tip might break, as is the case without any active damping. Different approaches based on feedback control are investigated; all implemented using an active boring bar. The first approach is based on a digital adaptive feedback controller; the feedback filtered-X LMS algorithm. The two other controllers are analog; one is a lead controller and the other is lead-lag controller, both with gain and phase orthogonally adjustable. This paper focuses on robustness, stability and convergence of the digital and the analog feedback controllers. The analysis is based on open loop frequency response function estimates during different operating conditions.

  • 483. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Developments Steps of an Active Boring Bar for Industrial Application2008Conference paper (Refereed)
    Abstract [en]

    In the manufacturing industry, boring bar vibration in internal turning is a frequent problem. Boring bars are often used to machine deep and precise holes in workpieces and are thus usually long and slender. The consequences of boring bar vibration are generally; reduced tool life, poor surface finish and high sound levels in the working environment. Boring bar vibration may however be attenuated using a suitable active vibration control system. Such system may for instance rely on feedback control and an active boring bar with embedded actuators. There are some steps between research results, having a controlled laboratory setup working, and applying the technology in industry, that are essential. Several prototypes of an active boring bar for industrial applications have been produced as part of this research. Different development phases and some challenges along the road to a successful product are discussed. Also, an active boring bar vibration control system in industry is presented and what challenges such a development may include.

  • 484. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Dynamic Properties of Tooling Structure: Hydraulic Clamping versus Standard Screw Clamping in a Lathe Application2008Conference paper (Other academic)
    Abstract [en]

    During turning, the metal deformation process excites the tooling structure and may result in high and degrading tool vibration levels. In internal turning the boring bar is generally the weakest link of the tooling structure. The boundary conditions imposed on the boring bar by the clamping housing influence the dynamic properties of a clamped boring bar. The dynamic properties of a clamped boring bar have been investigated for two different clamping housings; one that clamps the boring bar by means of hydraulic pressure and one that clamps the boring bar by means of clamping screws. Both experimental and analytical methods have been utilized in the investigation.

  • 485. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    INVESTIGATION OF THE DYNAMIC PROPERTIES OF A BORING BAR CONCERNING DIFFERENT BOUNDARY CONDITIONS2007Conference paper (Refereed)
    Abstract [en]

    The boring bar is one of the most widely used types of tool holder in metal cutting operations. The turning process subjects the tool to vibration, and cutting in deep workpiece cavities is likely to result in high vibration levels. The consequences of such vibration levels are generally; reduced tool life, poor surface finishing and disturbing sound. Internal turning frequently requires a long and slender boring bar in order to machine inside a cavity, and the vibrations generally become highly correlated with the fundamental bending mode of the boring bar. Different methods can be applied to reduce the vibrations, the implementation of the most efficient and stable methods require in depth knowledge concerning the dynamic properties of the tooling system. Furthermore, the interface between the boring bar and the clamping house has a significant influence on the dynamic properties of the clamped boring bar. In this paper different cases of boundary condition of the boring bar are presented partly analytically but also experimentally. This paper focuses on dynamic properties of a boring bar that arise due to different clamping conditions of the boring bar introduced by a clamping house commonly used in the manufacturing industry.

  • 486. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Lagö, Thomas L
    Vibration in Turning and the Active Control of Tool Vibration2007Conference paper (Refereed)
    Abstract [en]

    A frequent problem in the manufacturing industry today is the vibrations or chatter induced in machine tools during machining. Chatter in machine tools affect the result of machining, particularly the surface finish. Furthermore, tool life is correlated with the degree of vibration and acoustic noise introduced. The material deformation process during turning induces a broadband excitation of the machine tool and chatter or tool vibration is usually related to a low-order bending mode of, for example, the tool holder shank in external turning, the boring bar in internal turning. Frequently, the tool holder responses during continuous unstable machining have non-linear properties and exhibit advanced stochastic behaviour. Generally, the tooling structure - the interface between the cutting tool or insert and the machine tool - is the weakest link in a machining system. Active control of tool vibration, however, enables a flexible solution to selectively increase the dynamic stiffness of the tooling structure at the actual frequency of the dominating bending modes. Based on adequate controllers, tool holder shanks and boring bars with embedded actuators and vibration sensors tool vibration in turning may be reduced significantly.

  • 487. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Sigfridsson, Andreas
    Svensson, Tobias
    Lagö, Thomas L
    A First Prototype of an Active Boring Bar Tested in Industry2006Conference paper (Other academic)
    Abstract [en]

    Metal cutting is a common process in the manufacturing industry. Vibration problem during metal cutting is a reality for these manufactures. The vibration level depends on many different parameters such as material type, dimensions of the workpiece and boring bar, cutting data and operation mode. Internal cutting is one of the most troublesome operation modes, without any continuous monitoring and control from an operator grave vibration levels quickly arises. From the industry point of view this is an expense in the production line. The reduced tool life and the coarse surface finishing caused by the large vibration will force the operator to stop the cutting process, either to change tool when it is broken or change cutting data like decreasing the cutting depth. Any interference like this increases the working time for each component tremendous. These problems have been located and examined in an industry producing and renovating components using a lathe. One lathe was chosen for further investigation by experimental modal analysis and analysis during operational mode. The examination was done in an environment with machining processes in full operation mode and also during non-working ours. After analyzing the problem different solution where scrutinized and a first prototype where constructed. The solution to the problem is an active boring bar. This paper will present the procedure from analyzing a problem in industry to the test of the first prototype of the solution.

  • 488. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Håkansson, Lars
    Claesson, Ingvar
    Sigfridsson, Andreas
    Svensson, Tobias
    Lagö, Thomas L
    Active Boring Bar Prototype Tested in Industry2006Conference paper (Other academic)
    Abstract [en]

    Metal cutting processes are common in the manufacturing industry. Vibration problems during metal cutting are a degrading reality in the manufacturing industry and have a negative impact on the productivity and increase’s the production costs. The vibration level depends on many different parameters such as material type, dimensions of the workpiece, tool, tooling structure, machine structure, the cutting data and operation mode. Internal turning is one of the most troublesome operation modes concerning vibration. It generally requires that extra care must be taken with pro-duction planning and in the preparations for the machining of a workpiece. Frequently, continuous monitoring and control by a skilled operator is required to avoid grave degrading vibration levels. The reduced tool life and the coarse surface finishing generally caused by large vibration levels will e.g. force the operator to stop the cutting process, either to change tool when it is broken or change cutting data like decreasing the cutting depth. Any interference like this increases the production time and cost for each component tremendously. In an industry producing and renovating compo-nents using different machine tools, a particular lathe used for internal turning of certain details with extensive vibration problem was chosen for further investigation by experimental modal analysis and vibration analysis during operation. The examination was done in an environment with ma-chine tools in full operation and also during non-working ours. After analyzing the problem different solutions for the reduction of the vibration problem was investigated and subsequently a first proto-type was constructed. The solution to the problem is active control of the tool vibration using an active boring bar. This paper will present the procedure from analyzing a problem in industry to the test of the first prototype of the solution.

  • 489. Åkesson, Henrik
    et al.
    Smirnova, Tatiana
    Lagö, Thomas L
    Håkansson, Lars
    Analysis of Dynamic Properties of Boring Bars Concerning Different Clamping Conditions2007Report (Other academic)
    Abstract [en]

    The boring bar is one of the most widely used type of tool holders in metal cutting operations. The turning process subjects the tool to vibrations, and cutting in deep workpiece cavities is likely to result in high vibration levels. The consequences of such vibration levels generally results in: reduced tool life, poor surface finishing and disturbing sound. Internal turning frequently requires a long and slender boring bar in order to machine inside a cavity, and the vibrations generally become highly correlated with one of the fundamental bending modes of the boring bar. Different methods can be applied to reduce the vibrations, the implementation of the most efficient and stable methods require in depth knowledge concerning the dynamic properties of the tooling system. Furthermore, the interface between the boring bar and the clamping house has a significant influence on the dynamic properties of the clamped boring bar. This report focuses on the dynamic properties of a boring bar that arise under different clamping conditions of the boring bar and are introduced by a clamping house (commonly used in the manufacturing industry). The dynamic properties of a boring bar (for different cases of boundary condition of the boring bar) are presented partly analytically but also experimentally.

  • 490. Östlin, Erik
    On Radio Wave Propagation Measurements and Modelling for Cellular Mobile Radio Networks2009Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    To support the continuously increasing number of mobile telephone users around the world, mobile communication systems have become more advanced and sophisticated in their designs. As a result of the great success with the second generation mobile radio networks, deployment of the third and development of fourth generations, the demand for higher data rates to support available services, such as internet connection, video telephony and personal navigation systems, is ever growing. To be able to meet the requirements regarding bandwidth and number of users, enhancements of existing systems and introductions of conceptually new technologies and techniques have been researched and developed. Although new proposed technologies in theory provide increased network capacity, the backbone of a successful roll-out of a mobile telephone system is inevitably the planning of the network’s cellular structure. Hence, the fundamental aspect to a reliable cellular planning is the knowledge about the physical radio channel for wide sets of different propagation scenarios. Therefore, to study radio wave propagation in typical Australian environments, the Commonwealth Scientific and Industrial Research Organisation (CSIRO) and the Australian Telecommunications Cooperative Research Centre (ATcrc) in collaboration developed a cellular code division multiple access (CDMA) pilot scanner. The pilot scanner measurement equipment enables for radio wave propagation measurements in available commercial CDMA mobile radio networks, which in Australia are usually deployed for extensive rural areas. Over time, the collected measurement data has been used to characterise many different types of mobile radio environments and some of the results are presented in this thesis. The thesis is divided into an introduction section and four parts based on peer-reviewed international research publications. The introduction section presents the reader with some relevant background on channel and propagation modelling. Also, the CDMA scanner measurement system that was developed in parallel with the research results founding this thesis is presented. The first part presents work on the evaluation and development of the different revisions of the Recommendation ITU-R P.1546 point-to-area radio wave propagation prediction model. In particular, the modified application of the terrain clearance angle (TCA) and the calculation method of the effective antenna height are scrutinized. In the second part, the correlation between the smallscale fading characteristics, described by the Ricean K-factor, and the vegetation density in the vicinity of the mobile receiving antenna is investigated. The third part presents an artificial neural network (ANN) based technique incorporated to predict path loss in rural macrocell environments. Obtained results, such as prediction accuracy and training time, are presented for different sized ANNs and different training approaches. Finally, the fourth part proposes an extension of the path loss ANN enabling the model to also predict small-scale fading characteristics.

  • 491. Östlin, Erik
    et al.
    Suzuki, Hajime
    Zepernick, Hans-Jürgen
    Evaluation of the propagation model Recommendation ITU-R P. 1546 for mobile services in rural Australia2008Conference paper (Refereed)
    Abstract [en]

    In this paper, the validity of Recommendation ITU-R P.1546 in a short-range terrestrial environment is analyzed. Its three versions (P.1546, P.1546-1, and P.1546-2) are compared against simple models and evaluated using measurement results that were obtained by utilizing the pilot signal of a commercial code-division multiple-access mobile telephone network. Measurement results show that P.1546-2, on average, underestimates the field strength by more than 10 dB for typical Australian rural areas. However, it improves the error standard deviation compared to previous versions. The causes of these effects and the suggestions for further development of the Recommendation are discussed and evaluated.

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