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  • 1. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Applied Complex Chebyshev Optimization Using Dual Nested Complex Approximation2001Konferansepaper (Fagfellevurdert)
  • 2. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Nordholm, Sven
    Acoustic Echo Cancelling with microphone arrays1995Konferansepaper (Fagfellevurdert)
  • 3. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Nordholm, Sven
    Chebyshev Optimization of Circular Arrays Inequalities1998Konferansepaper (Fagfellevurdert)
  • 4. Dahl, Mattias
    Acoustic Noise and Echo Cancelling: Microphone Array Methods and Applications1997Licentiatavhandling, med artikler (Annet vitenskapelig)
    Abstract [en]

    This Licentiate thesis is divided into three parts corresponding to three different papers. There is one research report, one conference paper and one submitted journal paper. All three parts deal with acoustic echo and/or noise cancelling problems when using adaptive microphone arrays. In particular, the papers address the performance of an adaptive microphone array in a small enclosure such as the car cabin. A calibrating scheme is proposed which is independent of array geometry and channel matching, and which calibrates the adaptive array to the given acoustic environment and to the given electronic equipment. Results from real measurements in a car interior are included and compared with an analytical description of an adaptive microphone array. Part A gives an analytical description of an adaptive microphone array which facilitates a simple built-in calibration to the environment and instrumentation. Part B describes the method for performing acoustic echo cancelling with a dig-ital "on-site", "self-calibrating" microphone array system. The calibration process is a simple indirect calibration which continuously adapts to the actual environment and electronic equipment. There is a US patent based on this part and an international patent is currently under examination. Part C presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and µ-law.

  • 5. Dahl, Mattias
    Acoustic Noise and Echo Cancelling: Microphone Array Methods and Applications1997Rapport (Annet vitenskapelig)
    Abstract [en]

    This Licentiate thesis is divided into three parts corresponding to three different papers. There is one research report, one conference paper and one submitted journal paper. All three parts deal with acoustic echo and/or noise cancelling problems when using adaptive microphone arrays. In particular, the papers address the performance of an adaptive microphone array in a small enclosure such as the car cabin. A calibrating scheme is proposed which is independent of array geometry and channel matching, and which calibrates the adaptive array to the given acoustic environment and to the given electronic equipment. Results from real measurements in a car interior are included and compared with an analytical description of an adaptive microphone array. Part A gives an analytical description of an adaptive microphone array which facilitates a simple built-in calibration to the environment and instrumentation. Part B describes the method for performing acoustic echo cancelling with a dig-ital "on-site", "self-calibrating" microphone array system. The calibration process is a simple indirect calibration which continuously adapts to the actual environment and electronic equipment. There is a US patent based on this part and an international patent is currently under examination. Part C presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and µ-law.

  • 6. Dahl, Mattias
    Applied Array-Filter Design: Methods and Applications2000Doktoravhandling, med artikler (Annet vitenskapelig)
  • 7. Dahl, Mattias
    On-Site Calibrated Microphone Array for Mobile Communication2000Konferansepaper (Fagfellevurdert)
  • 8. Dahl, Mattias
    Speech Recognition in Severely Disturbed Environments Combining Ear-Mic and Active Noise Control2002Konferansepaper (Fagfellevurdert)
  • 9. Dahl, Mattias
    et al.
    Claesson, Ingvar
    A Neural Network Trained Microphone Array System for Noise Reduction1996Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and $mu@-law. Such a quantizer is designed to increase the Signal to Quantization Noise Ratio (SQNR) for small amplitudes in telecommunications systems. The proposed method primarily developed for hands-free mobile telephones, suppresses the ambient car noise with approximately 10 dB. The system is based upon a multi-layered nonlinear back-propagation trained network by using a built-in calibration technique.

  • 10. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Acoustic Echo Cancelling with Michrophone Arrays1995Rapport (Annet vitenskapelig)
    Abstract [en]

    This report describes a novel method to perform acoustic echo cancelling with microphone arrays. The method employs a digital self-calibrating microphone system. The on-site calibration process is a simple indirect calibration which adapts in each special case to the environment and electronic equipment. The method also continuously takes into account environmental disturbances such as car engine noise and fan noise. The method is primarily aimed at handsfree mobile telephones, by suppressing the handsfree loudspeaker and car noise simultaneously. The report also contains an extensive evaluation in a car.

  • 11. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Acoustic Noise and Echo Canceling with Microphone Array1999Inngår i: IEEE Transactions on Vehicular Technology , ISSN 0018-9545 , Vol. 48, nr 5, s. 1518-1526Artikkel i tidsskrift (Fagfellevurdert)
    Abstract [en]

    A novel method of performing acoustic echo canceling using microphone arrays is presented. The method employs a digital self-calibrating microphone system. The calibration process is a simple indirect on-site calibration that adapts to the particulars of the acoustic environment and the electronic equipment in use. Primarily intended for handsfree telephones in automobiles, the method simultaneously suppresses the handsfree loudspeaker and car noise. The system also continuously takes into account disturbances such as fan noise. Examples from an extensive evaluation in a car are also included. Typical performance results demonstrate 20-dB echo cancellation and 10-dB noise reduction simultaneously.

  • 12. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordebo, Sven
    Antenna Array Design using Dual Nested Complex Approximation2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents a new practical approach to complex Chebyshev approximation by semi-infinite linear programming. The approximation problem may be general with arbitrary complex basis functions. By the new front-end technique, the associated semi-infinite linear programming problem is solved exploiting the finiteness of the related Lagrange multipliers by adapting finite--dimensional linear programming to the dual semi--infinite problem, and thereby taking advantage of the numerical stability and efficiency of conventional linear programming software packages. Furthermore, the optimization procedure is simple to describe theoretically and straightforward to implement in computer coding. The new design technique is therefore highly accessible. The new algorithm is formally introduced as the linear Dual Nested Complex Approximation (DNCA) algorithm. The DNCA algorithm is versatile and can be applied to a variety of applications such as narrow-band as well as broad-band beamformers with any geometry, conventional Finite Impulse Response (FIR) filters, analog and digital Laguerre networks, and digital FIR equalizers. The proposed optimization technique is applied to several numerical examples dealing with the design of a narrow-band base-station antenna array for mobile communication. The flexibility and numerical efficiency of the proposed design technique are illustrated with these examples where hundreds of antenna elements are optimized without numerical difficulties.

  • 13. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordebo, Sven
    Simultaneous Echo Cancellation and Car Noise Suppression Employing a Microphone Array1997Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents a method to simultaneously perform 20 dB acoustic echo cancellation and 15-20 dB speech enhancement using an adaptive microphone array combined with spectral subtraction. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and the broadband car noise. The array system is based on a fast and efficient on-site calibration and can be used in other situations such as conventional speaker phones.

  • 14. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordebo, Sven
    Nordholm, Sven
    Chebyshev Optimization of Circular Arrays1999Konferansepaper (Fagfellevurdert)
  • 15. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordebo, Sven
    Nordholm, Sven
    En adaptiv mikrofon för bullerundertryckning i bil1994Konferansepaper (Fagfellevurdert)
  • 16. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordholm, Sven
    Grbic, Nedelko
    Adaptive Microphone Array System for Speech Enhancement1997Konferansepaper (Fagfellevurdert)
  • 17. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordholm, Sven
    Nordebo, Sven
    Acoustic Echo and Noise Cancelling using Microphone Arrays1996Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents a new method to simultaneously perform 20 dB acoustic echo cancellation and 10 dB speech enhancement utilizing an adaptive microphone array. The system is based on a fast and efficient on-site calibration. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and car noise. The method can also be used in other situations such as conventional speaker phones.

  • 18. Dahl, Mattias
    et al.
    Claesson, Ingvar
    Nordholm, Sven
    Nordebo, Sven
    Microphone Array and Acoustic Echo Canceller for Use in Hands-free Mobile Telephones1996Konferansepaper (Fagfellevurdert)
  • 19.
    Dahl, Mattias
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Javadi, Mohammad Saleh
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Analytical modelling for video-based vehicle speed measurement framework2019Inngår i: Optik (Stuttgart), ISSN 0030-4026, E-ISSN 1618-1336Artikkel i tidsskrift (Fagfellevurdert)
  • 20. Dahl, Mattias
    et al.
    Nordebo, Sven
    Claesson, Ingvar
    A New General Front-End Technique for Complex Quadratic Programming: Applications to Array Pattern Synthesis2000Rapport (Annet vitenskapelig)
    Abstract [en]

    This paper presents a new practical approach to complex quadratic programming which solves the broad class of complex approximation problems employing finitization of semi-infinite formulations. The approximation problem may be general with arbitrarily complex basis functions. By using a new technique, the associated semi-infinite quadratic programming problem can be solved taking advantage of the numerical stability and efficiency of conventional quadratic programming software packages. Furthermore, the optimization procedure is simple to describe theoretically and straightforward to implement in computer coding. The new design technique is therefore highly accessible. The complex approximation algorithm is versatile and can be applied to a variety of applications such as narrow-band as well as broad-band beamformers with any geometry, conventional FIR filters, digital Laguerre networks, and digital FIR equalizers. The new algorithm is formally introduced as the quadratic Dual Nested Complex Approximation (DNCA) algorithm. The essence of the new technique, justified by the Caratheodory's dimensionality theorem, is to exploit the finiteness of the related Lagrange multipliers by adapting conventional finite-dimensional quadratic programming to the semi-infinite quadratic programming re-formulation of complex approximation problems. The design criterion in our application is to minimize the side-lobe energy of an antenna array when subjected to a specified bound on the peak side-lobe level. Additional linear constraints are used to form the main-lobe. The design problem is formulated as a semi-infinite quadratic program and solved by using the new front-end applied on top of a software package for conventional finite-dimensional quadratic programming. The proposed optimization technique is applied to several numerical examples dealing with the design of a narrow-band base-station antenna array for mobile communication. The flexibility and numerical efficiency of the proposed design technique are illustrated with these examples where even hundreds of antenna elements are optimized without numerical difficulties.

  • 21. Dahl, Mattias
    et al.
    Nordebo, Sven
    Claesson, Ingvar
    Complex Approximation by Semi-Infinite Quadratic Programming2000Konferansepaper (Fagfellevurdert)
  • 22. Dahl, Mattias
    et al.
    Nordebo, Sven
    Claesson, Ingvar
    Complex Chebyshev Optimization Using Conventional Linear Programming: A versatile and comprehensive solution2000Rapport (Annet vitenskapelig)
    Abstract [en]

    This paper presents a new practical approach to semi-infinite complex Chebyshev approximation. By using a new technique, the general complex Chebyshev approximation problem can be solved with arbitrary base functions taking advantage of the numerical stability and efficiency of conventional linear programming software packages. Furthermore, the optimization procedure is simple to describe theoretically and straightforward to implement in computer coding. The new design technique is therefore highly accessible. The complex approximation algorithm is general and can be applied to a variety of applications such as conventional FIR filters, narrow-band as well as broad-band beamformers with any geometry, the digital Laguerre networks, and digital FIR equalizers. The new algorithm is formally introduced as the Dual Nested Complex Approximation (DNCA) linear programming algorithm. The design example in limelight is array pattern synthesis of a mobile base-station antenna array. The corresponding design formulation is general and facilitates treatment of the solution of problems with arbitrary array geometry and side-lobe weighting. The complex approximation problem is formulated as a semi-infinite linear program and solved by using a front-end applied on top of a software package for conventional finite-dimensional linear programming. The essence of the new technique, justified by the Caratheodory dimensionality theorem, is to exploit the finiteness of the related Lagrange multipliers by adapting conventional finite-dimensional linear programming to the semi-infinite linear programming problem. The proposed optimization technique is applied to several numerical examples dealing with the design of a narrow-band base-station antenna array for mobile communication. The flexibility and numerical efficiency of the proposed design technique are illustrated with these examples where even hundreds of antenna elements are optimized without numerical difficulties.

  • 23. Dahl, Mattias
    et al.
    Tran, To
    Claesson, Ingvar
    Nordebo, Sven
    Design of antenna array using dual nested complex approximation2005Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents a new practical approach to complex Chebyshev approximation by semi-infinite linear programming. By the new front-end technique, the associated semi-infinite linear programming problem is solved exploiting the finiteness of the related Lagrange multipliers by adapting finite-dimensional linear programming to the dual semi-infinite problem, and thereby taking advantage of the numerical siability and efficiency of conventional linear programming software packages. Furthermore, the optimization procedure is simple to describe theoretically and straightforward to implement in computer coding. The new design technique is therefore. highly accessible. The algorithm is formally introduced as the linear Dual Nested Complex Approximation (DNCA) algorithm. The DNCA algorithm is versatile and can be applied to a variety of applications such as narrow-band as well as broad-band beamformers with any geometry, conventional Finite Impulse Response (FIR) filters, analog and digital Laguerre networks. and digital FIR equalizers. The proposed optimization technique is applied to several numerical examples dealing with the design of a narrow-band base-station antenna array for mobile communication.

  • 24. Grbic, Nedelko
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Acoustic Echo Cancelling and Noise Suppression with Microphone Arrays1999Rapport (Annet vitenskapelig)
    Abstract [en]

    This report presents a method to achieve acoustic echo canceling and noise suppression using microphone arrays. The method employs a digital self-calibrating microphone system. The on-site calibration process is a simple indirect calibration which adapts in each specific case to the environment and the electronic equipment used. The method also continuously reduces environmental disturbances such as car engine noise and fan noise. The method is primarily aimed at hands free mobile telephones by suppressing the hands free loudspeaker and car cabin noise simultaneously. The report also contains an evaluation of the impact of echo and noise suppression on a real conversation, accomplished in a car using a microphone array.

  • 25. Grbic, Nedelko
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Neural Network Based Adaptive Microphone Array System for Speech Enhancement1998Konferansepaper (Fagfellevurdert)
    Abstract [en]

    Presents a microphone array system for use in handsfree mobile telephone equipment. The array is based on a fast and efficient “on-site” and “self- calibration” scheme. The performance in suppressing the interior car cabin noise and the far-end speech is approximately 17 dB, respectively, while maintaining the near-end speaker level. The near-end signal is almost undistorted. The performance of two different algorithms, normalized least-mean-square (NLMS) and fully connected backpropagation supervised neural network (MLP-NN) are evaluated. The proposed microphone array calibration scheme can also be used in other situations such as speech recognition devices.

  • 26.
    Holmgren, Johan
    et al.
    Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, Avdelningen för matematik och naturvetenskap.
    Davidsson, Paul
    Person, Jan A.
    Agent-based simulation of freight transport between geographical zones2013Konferansepaper (Fagfellevurdert)
    Abstract [en]

    We present TAPAS-Z, which is an agent-based freight transport analysis model for simulation of decision-making and transport activities. TAPAS-Z is a further development of a simulation model called TAPAS, and it has improved support for simulation of transport in large geographical regions. It is based on the principles that shipments are simulated for chosen supplier-consumer relations in a geographic region, and that the geographic locations of suppliers and consumers are randomly varied for each shipment. In TAPAS-Z, one supplier represents all real-world suppliers in a geographic zone, and one consumer represents all real-world consumers in a zone. In that way, TAPAS-Z is able to capture some of the diversity in freight transport that is caused by the varying geographic locations of senders and receivers, and which is important when assessing the impact of transport policy and infrastructural measures.

  • 27. Håkansson, Lars
    et al.
    Johansson, Sven
    Dahl, Mattias
    Sjösten, Per
    Claesson, Ingvar
    NOISE CANCELLING HEADSETS FOR SPEECH COMMUNICATION2002Inngår i: Noise Reduction in Speech Applications / [ed] Davis, Gillian M., Boca Raton, Florida: CRC Press , 2002, s. 305-327Kapittel i bok, del av antologi (Annet vitenskapelig)
    Abstract [en]

    Headsets for speech communication are used in a wide range of applications. The basic idea is to allow hands-free speech communication, leaving both hands available for other tasks. One typical headset application is aircraft pilot communication. The pilot must be able to communicate with personnel on the ground and at the same time use both hands to control the aircraft. Communication headsets usually consist of a pair of headphones and a microphone attached with an adjustable boom. Headphone design varies widely between different manufacturers and models. In its simplest form, the headphone has an open construction providing little or no attenuation of the environmental noise. In headsets designed for noisy environments, the headphones are mounted in ear cups with cushions that provide some attenuation. The microphone is primarily designed to pick up the speech signal, but if the headset is used in a noisy environment, the background noise will also be picked up and transmitted together with the speech. As a consequence, speech intelligibility at the receive end will be reduced, possibly to zero. To increase the speech-to-noise ratio, it is common to use a directional microphone that has a lower sensitivity to sound incident from other directions than the frontal direction. In addition to this, the microphone electronics are usually equipped with a gate function that completely shuts off the microphone signal if its level drops below a threshold value. The purpose of the gate is to open the channel for transmission only when a speech signal is present. Headsets are frequently used in noisy environments where they suffer from problems of speech intelligibility. Even if an ear-cup type headset is used, the attenuation is relatively poor for low frequencies. Low frequency noise has a masking effect on speech, which significantly reduces the speech intelligibility. Several cases have been reported in which the sound level of the communication signal was increased to hazardous levels by the user to overcome this low frequency masking effect [1,2]. Ear exposure to the communication system resulted in hearing damage, such as hearing loss, tinnitus and hyperacusis.

  • 28.
    Javadi, Mohammad Saleh
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Pettersson, Mats
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Change detection in aerial images using a Kendall's TAU distance pattern correlation2016Inngår i: PROCEEDINGS OF THE 2016 6TH EUROPEAN WORKSHOP ON VISUAL INFORMATION PROCESSING (EUVIP), IEEE, 2016Konferansepaper (Fagfellevurdert)
    Abstract [en]

    Change detection in aerial images is the core of many remote sensing applications to analyze the dynamics of a wide area on the ground. In this paper, a remote sensing method is proposed based on viewpoint transformation and a modified Kendall rank correlation measure to detect changes in oblique aerial images. First, the different viewpoints of the aerial images are compromised and then, a local pattern descriptor based on Kendall rank correlation coefficient is introduced. A new distance measure referred to as Kendall's Tau-d (Tau distance) coefficient is presented to determine the changed regions. The developed system is applied on oblique aerial images with very low aspect angles that obtained using an unmanned aerial vehicle in two different days with drastic change in illumination and weather conditions. The experimental results indicate the robustness of the proposed method to variant illumination, shadows and multiple viewpoints for change detection in aerial images.

  • 29.
    Javadi, Mohammad Saleh
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Pettersson, Mats
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Vehicle speed measurement model for video-based systems2019Inngår i: Computers & electrical engineering, ISSN 0045-7906, E-ISSN 1879-0755, Vol. 76, s. 238-248Artikkel i tidsskrift (Fagfellevurdert)
    Abstract [en]

    Advanced analysis of road traffic data is an essential component of today's intelligent transportation systems. This paper presents a video-based vehicle speed measurement system based on a proposed mathematical model using a movement pattern vector as an input variable. The system uses the intrusion line technique to measure the movement pattern vector with low computational complexity. Further, the mathematical model introduced to generate the pdf (probability density function) of a vehicle's speed that improves the speed estimate. As a result, the presented model provides a reliable framework with which to optically measure the speeds of passing vehicles with high accuracy. As a proof of concept, the proposed method was tested on a busy highway under realistic circumstances. The results were validated by a GPS (Global Positioning System)-equipped car and the traffic regulations at the measurement site. The experimental results are promising, with an average error of 1.77 % in challenging scenarios.

  • 30.
    Javadi, Mohammad Saleh
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Pettersson, Mats
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Kulesza, Wlodek
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Design of a video-based vehicle speed measurement system: an uncertainty approach2019Inngår i: 2018 Joint 7th International Conference on Informatics, Electronics & Vision (ICIEV) and 2018 2nd International Conference on Imaging, Vision & Pattern Recognition (icIVPR), Kitakyushu, Japan, 2018, pp. 44-49., IEEE, 2019, artikkel-id 8640964Konferansepaper (Fagfellevurdert)
    Abstract [en]

    Speed measurement is one of the key components of intelligent transportation systems. It provides suitable information for traffic management and law enforcement. This paper presents a versatile and analytical model for a video-based speed measurement in form of the probability density function (PDF). In the proposed model, the main factors contributing to the uncertainties of the measurement are considered. Furthermore, a guideline is introduced in order to design a video-based speed measurement system based on the traffic and other requirements. As a proof of concept, the model has been simulated and tested for various speeds. An evaluation validates the strength of the model for accurate speed measurement under realistic circumstances.

  • 31.
    Javadi, Mohammad Saleh
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Rameez, Muhammad
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Pettersson, Mats
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap.
    Vehicle classification based on multiple fuzzy c-means clustering using dimensions and speed features2018Inngår i: Procedia Computer Science, Elsevier, 2018, Vol. 126, s. 7s. 1344-1350Konferansepaper (Fagfellevurdert)
    Abstract [en]

    Vehicle classification has a significant use in traffic surveillance and management. There are many methods proposed to accomplish this task using variety of sensorS. In this paper, a method based on fuzzy c-means (FCM) clustering is introduced that uses dimensions and speed features of each vehicle. This method exploits the distinction in dimensions features and traffic regulations for each class of vehicles by using multiple FCM clusterings and initializing the partition matrices of the respective classifierS. The experimental results demonstrate that the proposed approach is successful in clustering vehicles from different classes with similar appearanceS. In addition, it is fast and efficient for big data analysiS.

  • 32. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Computational Efficient Method for Assuring Full Duplex Feeling in Hands-free Communication2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This report proposes a method for obtaining satisfying "full-duplex feeling" in hands-free communication units at low computational cost. The proposed method uses a combination of an acoustic echo cancellation unit and an adaptive gain unit. The core of the method is to perform the processing of the speech signal into two separate frequency bands and to process these in different manners. Acoustic echoes in the low frequency part of the signal are cancelled by means of an acoustic echo cancellation unit, while acoustic echoes in the high frequency part are suppressed by an adaptive gain unit.

  • 33. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Computational Efficient Method for Assuring Full Duplex Feeling in Hands-free Communication2003Rapport (Annet vitenskapelig)
    Abstract [en]

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

  • 34. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A computational Efficient Method for Assuring Full-Duplex Felling in Hands-Free Communication2002Konferansepaper (Fagfellevurdert)
  • 35. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Computational Efficient Method For Bandwidth Extension of a Conference Phone2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 36. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A Finite Precision LMS Algorithm for Increased Quantization Robustness2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 37. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    An open-loop doubletalk detector using power spectrum estimation2004Konferansepaper (Fagfellevurdert)
  • 38. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Delayed Filter Update: An Acoustic Echo Canceler Structure for Improved Doubletalk Detection2003Konferansepaper (Fagfellevurdert)
  • 39. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    On audio hands-free design2004Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 40. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    The two-path algorithm for line echo cancellation2004Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 41. Lindström, Fredric
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Stuns, S
    An LMS Based Algorithm for reduced Finite Precision Effects2002Konferansepaper (Fagfellevurdert)
  • 42. Lindström, Fredric
    et al.
    Eriksson, John-Erik
    Dahl, Mattias
    Claesson, Ingvar
    On the Design of a Sound System for a Mobile Audio Unit2005Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 43. Lindström, Fredric
    et al.
    Schüldt, Christian
    Dahl, Mattias
    Claesson, Ingvar
    Improving the Performance of a Low-Complexity Doubletalk2005Konferansepaper (Fagfellevurdert)
  • 44.
    Mbiydzenyuy, Gideon
    et al.
    Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation.
    Dahl, Mattias
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, Avdelningen för matematik och naturvetenskap.
    Holmgren, Johan
    Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation.
    Road travel time prediction: A micro-level sampling approach2013Konferansepaper (Fagfellevurdert)
    Abstract [en]

    The ability to generate accurate travel time predictions for road freight transport is important when, for example, estimating the arrival times for heavy goods vehicles (HGVs) in order to plan terminal activities. We present a micro-level sampling method for road travel time prediction. The method makes use of historical GPS-data in order to determine the movement of a vehicle from an origin to a destination along a specific route. The method generates a travel time distribution, which can be used to obtain the expected travel time and probabilities for deviations. The method has been illustrated and evaluated in an experiment where the effective travel time was predicted for transport between two terminals. The experiment made use of GPS data that was recorded for two HGVs during a period of two months. An important feature of the method is that it does not need road network information, such as speed limits and number of lanes.

  • 45. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    A cepstrum domain HMM-based speech enhancement method applied to nonstationary noise2005Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 46. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Digital Filter Design of IIR Filters using Real Valued Genetic Algorithm2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 47. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Gray-Scale Image Enhancement using the SMQT2005Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper explores the Successive Mean Quantization Transform (SMQT) for automatic enhancement of gray-scale images. The transform is in the paper presented using set theory. The image enhancement capabilities and properties of the transform are analyzed. The transform is capable to perform both a nonlinear and a shape preserving stretch of the image histogram. Experiments and comparisons to histogram equalization are conducted.

  • 48. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    HMM-based speech enhancement applied in non-stationary noise using cepstral features and log-normal approximation2003Konferansepaper (Fagfellevurdert)
    Abstract [en]

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

  • 49. Nilsson, Mikael
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    The Successive Mean Quantization Transform2005Konferansepaper (Fagfellevurdert)
    Abstract [en]

    This paper presents the Successive Mean Quantization Transform (SMQT). The transform reveals the organization or structure of the data and removes properties such as gain and bias. The transform is described and applied in speech processing and image processing. The SMQT is considered as an extra processing step for the mel frequency cepstral coefficients commonly used in speech recognition. In image processing the transform is applied in automatic image enhancement and dynamic range compression.

  • 50. Nordberg, Jörgen
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Nordholm, Sven
    Nordebo, Sven
    Acoustic Echo Cancellation Employing Delayless Subband Adaptive Filters1996Konferansepaper (Fagfellevurdert)
    Abstract [en]

    The use of hands-free communication in cars, computer applications and video conferencing has created a demand for high-quality acoustic echo cancellation. In these applications these acoustic channel has typically a long impulse response in the order of 100ms. Typical lengths of adaptive FIR-filters can be 500-1500 taps. In order to reduce the complexity and also to improve the convergence rate, subband processing schemes have been suggested. This paper presents an implementation of a delayless subband adaptive filter. The study shows a possible suppression of about 30 dB and also a more rapid convergence compared to a fullband LMS-filter.

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