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Claesson, Ingvar
Publications (10 of 361) Show all publications
Gertsovich, I., Nilsson, M., Ström Bartunek, J. & Claesson, I. (2018). Automatic estimation of a scale resolution in forensic images. Forensic Science International, 283, 58-71
Open this publication in new window or tab >>Automatic estimation of a scale resolution in forensic images
2018 (English)In: Forensic Science International, ISSN 0379-0738, E-ISSN 1872-6283, Vol. 283, p. 58-71Article in journal (Refereed) Published
Abstract [en]

This paper proposes a new method for an automatic detection of a resolution of a scale or a ruler with graduation marks in the shoeprint images. The method creates a vector of the correlations estimated from the co-occurrence matrices for every row in a shoeprint image. The scale resolution is estimated from maxima in Fourier spectrum of the correlations’ vectors. The proposed method is evaluated on over 500 images taken at crime scenes and in a forensics laboratory. The experimental results indicate the possibility of applying the proposed method to automatically estimate the scale resolution in forensic images. The automatic detection of a scale resolution could be used to automatically rescale a forensic image before the printing this image in “one-to-one” scale. Furthermore, the proposed method could be used to automatically rescale images to an equal scale thus allowing to compare the images digitally. © 2017 Elsevier B.V.

Place, publisher, year, edition, pages
Elsevier Ireland Ltd, 2018
Keywords
Gray level co-occurrence matrix, Near regular texture, Scale resolution estimation, Shoeprint image, Texture pattern periodicity
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-15713 (URN)10.1016/j.forsciint.2017.12.007 (DOI)000424296400013 ()
Available from: 2018-01-04 Created: 2018-01-04 Last updated: 2018-02-22Bibliographically approved
Yiu, K., Gao, M., Shiu, T., Wu, S., Tran, T. Q. & Claesson, I. (2013). A fast algorithm for the optimal design of high accuracy windows in signal processing. Optimization Methods and Software, 28(4), 900-916
Open this publication in new window or tab >>A fast algorithm for the optimal design of high accuracy windows in signal processing
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2013 (English)In: Optimization Methods and Software, ISSN 1055-6788, E-ISSN 1029-4937, Vol. 28, no 4, p. 900-916Article in journal (Refereed) Published
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.

Place, publisher, year, edition, pages
Taylor & Francis, 2013
Keywords
window function, filter design
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-6898 (URN)10.1080/10556788.2012.681659 (DOI)000320925500015 ()oai:bth.se:forskinfo920B8A8D519F1DD2C1257BE2003FD6DB (Local ID)oai:bth.se:forskinfo920B8A8D519F1DD2C1257BE2003FD6DB (Archive number)oai:bth.se:forskinfo920B8A8D519F1DD2C1257BE2003FD6DB (OAI)
Available from: 2013-09-10 Created: 2013-09-10 Last updated: 2017-12-04Bibliographically approved
Bartunek, J. S., Nilsson, M., Sällberg, B. & Claesson, I. (2013). Adaptive Fingerprint Image Enhancement With Emphasis on Preprocessing of Data. IEEE Transactions on Image Processing, 22(2), 644-656
Open this publication in new window or tab >>Adaptive Fingerprint Image Enhancement With Emphasis on Preprocessing of Data
2013 (English)In: IEEE Transactions on Image Processing, ISSN 1057-7149, E-ISSN 1941-0042, Vol. 22, no 2, p. 644-656Article in journal (Refereed) Published
Abstract [en]

This article proposes several improvements to an adaptive fingerprint enhancement method that is based on contextual filtering. The term adaptive implies that parameters of the method are automatically adjusted based on the input fingerprint image. Five processing blocks comprise the adaptive fingerprint enhancement method, where four of these blocks are updated in our proposed system. Hence, the proposed overall system is novel. The four updated processing blocks are: 1) preprocessing; 2) global analysis; 3) local analysis; and 4) matched filtering. In the preprocessing and local analysis blocks, a nonlinear dynamic range adjustment method is used. In the global analysis and matched filtering blocks, different forms of order statistical filters are applied. These processing blocks yield an improved and new adaptive fingerprint image processing method. The performance of the updated processing blocks is presented in the evaluation part of this paper. The algorithm is evaluated toward the NIST developed NBIS software for fingerprint recognition on FVC databases.

Place, publisher, year, edition, pages
IEEE, 2013
Keywords
Directional filtering, Fourier transform, image processing, spectral feature estimation, successive mean quantization transform
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7002 (URN)10.1109/TIP.2012.2220373 (DOI)000314717800019 ()oai:bth.se:forskinfoB04EDCB08DEC540DC1257B2F003ADC77 (Local ID)oai:bth.se:forskinfoB04EDCB08DEC540DC1257B2F003ADC77 (Archive number)oai:bth.se:forskinfoB04EDCB08DEC540DC1257B2F003ADC77 (OAI)
External cooperation:
Available from: 2013-03-18 Created: 2013-03-15 Last updated: 2017-12-04Bibliographically approved
Borgh, M., Schüldt, C. & Claesson, I. (2013). Efficient asynchronous re-sampling implementation on a low-power fixed-point DSP. In: : . Paper presented at IEEE International Conference on Consumer Electronics, ICCE. Las Vegas: IEEE
Open this publication in new window or tab >>Efficient asynchronous re-sampling implementation on a low-power fixed-point DSP
2013 (English)Conference paper, Published paper (Refereed)
Abstract [en]

This paper presents an asynchronous resampling implementation on a low-power fixed-point DSP, which uses around 47% less computational resources compared to the solution provided by the DSP manufacturer, without compromising audio quality.

Place, publisher, year, edition, pages
Las Vegas: IEEE, 2013
Keywords
Audio quality, Computational resources, Fixed-point DSP, Low Power, Resampling
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-6981 (URN)10.1109/ICCE.2013.6486937 (DOI)000318797800169 ()oai:bth.se:forskinfoA73140CC34227255C1257B7500457A25 (Local ID)978-1-4673-1363-6 (ISBN)oai:bth.se:forskinfoA73140CC34227255C1257B7500457A25 (Archive number)oai:bth.se:forskinfoA73140CC34227255C1257B7500457A25 (OAI)
Conference
IEEE International Conference on Consumer Electronics, ICCE
Available from: 2013-05-24 Created: 2013-05-24 Last updated: 2018-01-17Bibliographically approved
Khan, I., Gertsovich, I., Claesson, I., Håkansson, L., Johansson, P.-E., Wirenstedt, M., . . . Petersson, S. (2013). MRI SCANNER VIBRATION PATH ANALYSIS. In: Machinery Noise and Vibration: . Paper presented at The 20th International Congress on Sound and Vibration (ICSV20), Bangkok. , Article ID 725.
Open this publication in new window or tab >>MRI SCANNER VIBRATION PATH ANALYSIS
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2013 (English)In: Machinery Noise and Vibration, 2013, article id 725Conference paper, Published paper (Other academic)
Abstract [en]

Magnetic Resonance Imaging (MRI) scanner is one of the most important tools in clinical diagnostics. MRI scanners are associated by strong vibration which results in unpleasant and disturbing acoustic noise. The primary source of this vibration is the Lorentz force produced by fast switching of the currents inside the gradient coils of MRI scanners under a strong static magnetic field. During an MR-imaging scan the switching is controlled in order to spatially code the hydrogen nuclei that will generate the signal, which is reconstructed into anatomical images. Faster switching of the currents allows for shorter scan times and/or higher image resolutions. Consequently, the clinical quality has motivated the drive for shorter switching time and higher currents. This development, however, has also caused an undesired increase of MRI vibrations. The overall vibration phenomenon of an installed fully functional MRI scanner system becomes unique because of the installed location and ambiance. This vibration can potentially degrade the image quality and hence the diagnosis. Apart from the vibration produced, the associated annoying acoustic noise may not only affect the patients under examination and the clinical staff, but may also be transmitted to other parts of the building and causing discomfort for the personnel working there. In order to devise an effective isolation plan or improve an existing one both for vibration and acoustic noise it is important to study the noise and vibration transfer paths. This paper concerns an investigation of vibration transfer paths for vibration excited by an installed functional MRI scanner at a medical facility. The vibration transfer paths have been investigated experimentally. The obtained results are presented and discussed.

National Category
Signal Processing
Identifiers
urn:nbn:se:bth-13808 (URN)
Conference
The 20th International Congress on Sound and Vibration (ICSV20), Bangkok
Available from: 2017-01-22 Created: 2017-01-22 Last updated: 2018-05-22Bibliographically approved
Khan, M. G., Sällberg, B., Nordberg, J., Tufvesson, F. & Claesson, I. (2013). Non-Coherent Fourth-Order Detector for Impulse Radio Ultra Wideband Systems: Empirical Evaluation Using Channel Measurements. Wireless personal communications, 68(1), 27-46
Open this publication in new window or tab >>Non-Coherent Fourth-Order Detector for Impulse Radio Ultra Wideband Systems: Empirical Evaluation Using Channel Measurements
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2013 (English)In: Wireless personal communications, ISSN 0929-6212, E-ISSN 1572-834X, Vol. 68, no 1, p. 27-46Article in journal (Refereed) Published
Abstract [en]

Low-complex and low-power non-coherent energy detectors (EDs) are interesting for low data rate impulse radio (IR) ultra wideband (UWB) systems, but suffer from a loss in performance compared to coherent receivers. The performance of an ED also strongly depends on the integration interval (window size) of the integrator and the window position. This paper presents a non-coherent fourth-order detector (FD) which can discriminate between Gaussian noise signals and non-Gaussian IR-UWB signals by directly estimating the fourth-order moment of the received signal. The performance of the detectors is evaluated using realistic channels measured in a corridor, an office and a laboratory environment. The results show that bit-error-rate (BER) performance of the proposed FD receiver is slightly better than the ED in low signal-to-noise ratio (SNR) region and its performance improves as the SNR increases. In addition, BER of the FD receiver is less sensitive to overestimation of the integration interval making it relatively robust to variations of the channel delay spread. Finally, a criteria for the selection of integration time of the proposed detector is suggested.

Place, publisher, year, edition, pages
Springer, 2013
Keywords
Channel measurements, Higher-order moments, Non-coherent detection, UWB communications
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-6995 (URN)10.1007/s11277-011-0437-x (DOI)000314089600003 ()oai:bth.se:forskinfo067927955D2A3B39C12579740048441F (Local ID)oai:bth.se:forskinfo067927955D2A3B39C12579740048441F (Archive number)oai:bth.se:forskinfo067927955D2A3B39C12579740048441F (OAI)
Available from: 2013-05-07 Created: 2011-12-28 Last updated: 2017-12-04Bibliographically approved
Schüldt, C., Lindström, F. & Claesson, I. (2012). A Delay-Based Double-Talk Detector. IEEE Transactions on Audio, Speech, and Language Processing, 20(6), 1725-1733
Open this publication in new window or tab >>A Delay-Based Double-Talk Detector
2012 (English)In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 20, no 6, p. 1725-1733Article in journal (Refereed) Published
Abstract [en]

When an adaptive filter is used for echo cancellation, it is essential to prevent the filter from diverging in situations when the echo signal is contaminated with near-end disturbance, i.e. during double-talk. This paper presents an extension of a previously proposed double-talk detector for improved performance. It is shown that the computational complexity of the proposed detector is lower than that of the well-used normalized cross correlation (NCC) double-talk detector, at the cost of performance. Further, it is shown that there can be a significant performance difference, in terms of detecting double-talk, between having a fixed echo cancellation filter, which is a common strategy in objective evaluation techniques, and an adaptive filter, which is more close to realistic conditions.

Keywords
Echo cancellation, adaptive filters, double-talk, double-talk detection
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7170 (URN)10.1109/TASL.2012.2188512 (DOI)000302532000005 ()oai:bth.se:forskinfoC8E2C4C849EAE429C12579F700565253 (Local ID)oai:bth.se:forskinfoC8E2C4C849EAE429C12579F700565253 (Archive number)oai:bth.se:forskinfoC8E2C4C849EAE429C12579F700565253 (OAI)
External cooperation:
Note

http://ieeexplore.ieee.org/xpl/articleDetails.jsp?arnumber=6155602

Available from: 2012-11-27 Created: 2012-05-07 Last updated: 2017-12-04Bibliographically approved
Ishaq, R., Shahid, M., Lövström, B., Zapirain, B. G. & Claesson, I. (2012). Modulation frequency domain adaptive gain equalizer using convex optimization. Paper presented at 6th International Conference on Signal Processing and Communication Systems (ICSPCS). Paper presented at 6th International Conference on Signal Processing and Communication Systems (ICSPCS). Gold Coast, Australia: IEEE
Open this publication in new window or tab >>Modulation frequency domain adaptive gain equalizer using convex optimization
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2012 (English)Conference paper, Published paper (Refereed) Published
Abstract [en]

Adaptive gain equalizer (AGE) is a commonly used single-channel speech enhancement algorithm. AGE and its variants has been widely used for speech enhancement applications. There are two broad categories of these variants. The first deals with its improvement in time-frequency domain with readjustment of the used parameters and the second one deals with performing the main filtering operation in modulation frequency domain. This paper evaluates the working of AGE in modulation frequency domain with the use of a demodulation technique which solves the demodulation process as a convex optimization problem. The performance of the modified AGE is compared with the traditional AGE and another modulation frequency domain AGE based on demodulation using the spectral center-of-gravity. These used performance measures are Signal to Noise Ratio Improvement(SNRI), Spectral Distortion(SD) and Mean Option Score(MOS).

Place, publisher, year, edition, pages
Gold Coast, Australia: IEEE, 2012
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-6857 (URN)10.1109/ICSPCS.2012.6508019 (DOI)oai:bth.se:forskinfoDBF28EE2381414B0C1257C29006D27DE (Local ID)978-1-4673-2391-8 (ISBN)oai:bth.se:forskinfoDBF28EE2381414B0C1257C29006D27DE (Archive number)oai:bth.se:forskinfoDBF28EE2381414B0C1257C29006D27DE (OAI)
Conference
6th International Conference on Signal Processing and Communication Systems (ICSPCS)
Available from: 2013-11-22 Created: 2013-11-20 Last updated: 2015-06-30Bibliographically approved
Schüldt, C., Lindström, F. & Claesson, I. (2012). Robust low-complexity transfer logic for two-path echo cancellation. In: : . Paper presented at IEEE International Conference on Acoustics, Speech, and Signal Processing. Kyoto: IEEE
Open this publication in new window or tab >>Robust low-complexity transfer logic for two-path echo cancellation
2012 (English)Conference paper, Published paper (Refereed)
Abstract [en]

A well used approach for echo cancellation is the two-path method, where two adaptive filters in parallel are utilized. Typically, one filter is continuously updated, and when this filter is considered better adjusted to the echo-path than the other filter, the coefficients of the better adjusted filter is transferred to the other filter. When this transfer should occur is controlled by the transfer logic. This paper proposes transfer logic that is both more robust and more simple to tune, owing to fewer parameters, than the conventional approach. Extensive simulations show the advantages of the proposed method.

Place, publisher, year, edition, pages
Kyoto: IEEE, 2012
Keywords
Echo cancellation, adaptive filters, two-path, transfer logic
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7299 (URN)oai:bth.se:forskinfoC59323AFC0C70FF0C12579F70055314A (Local ID)978-1-4673-0046-9 (ISBN)oai:bth.se:forskinfoC59323AFC0C70FF0C12579F70055314A (Archive number)oai:bth.se:forskinfoC59323AFC0C70FF0C12579F70055314A (OAI)
External cooperation:
Conference
IEEE International Conference on Acoustics, Speech, and Signal Processing
Available from: 2012-09-18 Created: 2012-05-07 Last updated: 2016-09-06Bibliographically approved
Larsson, M., Nilsson, K., Johansson, S., Claesson, I. & Håkansson, L. (2011). An Active Noise Control Approach for Attenuating Noise Above the Plane Wave Region in Ducts. Paper presented at International Congress on Sound and Vibration, ICSV. Paper presented at International Congress on Sound and Vibration, ICSV. Rio de Janeiro, Brazil: International Institute of Acoustics and Vibration (IIAV)
Open this publication in new window or tab >>An Active Noise Control Approach for Attenuating Noise Above the Plane Wave Region in Ducts
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2011 (English)Conference paper, Published paper (Refereed) Published
Abstract [en]

In a narrow duct, a relatively simple single-channel feedforward ANC system may be used to attenuate noise propagating as plane waves. However, for ducts with larger dimensions the cut-on frequencies for one or several higher-order acoustic modes may be within the frequency range where ANC is applied. In such situations it is generally necessary to use a multiplechannel feedforward ANC system with several secondary sources, error sensors, and perhaps reference sensors. Such a system has a significantly higher complexity than a single-channel ANC system. In this paper another approach is described. Instead of using a multiple-channel feedforward ANC system on a duct of large dimension, the idea is to divide the duct into several more narrow parallel ducts. In this way the complexity of the ANC system may be reduced. In the experiments conducted for this paper, a duct was divided into two more narrow ducts. The noise propagating in each duct was controlled by a feedforward ANC system based on the leaky filtered-x LMS algorithm, where different reference- and error microphone configurations were used. The different configurations were compared to a configuration where the noise in respective narrow duct was controlled using a basic single-channel ANC system per duct. The results preliminary show that high attenuation of low-frequency noise in a duct of large dimension may be achieved using this approach.

Place, publisher, year, edition, pages
Rio de Janeiro, Brazil: International Institute of Acoustics and Vibration (IIAV), 2011
Keywords
Active Noise Control, ANC, Ducts, Plane Wave, Higher order modes
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7480 (URN)oai:bth.se:forskinfoDA3CFFAF64112ACAC12578ED002E6AC0 (Local ID)oai:bth.se:forskinfoDA3CFFAF64112ACAC12578ED002E6AC0 (Archive number)oai:bth.se:forskinfoDA3CFFAF64112ACAC12578ED002E6AC0 (OAI)
Conference
International Congress on Sound and Vibration, ICSV
Available from: 2012-09-18 Created: 2011-08-15 Last updated: 2015-06-30Bibliographically approved
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