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Träfflista för sökning "hsv:(TEKNIK OCH TEKNOLOGIER) hsv:(Elektroteknik och elektronik) ;pers:(Claesson Ingvar)"

Search: hsv:(TEKNIK OCH TEKNOLOGIER) hsv:(Elektroteknik och elektronik) > Claesson Ingvar

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1.
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2.
  • Nordholm, Sven, et al. (author)
  • Performance Limits in Subband Beamforming
  • 2003
  • In: IEEE transactions on speech and audio processing. - : IEEE. - 1063-6676 .- 1558-2353 .- 2329-9290. ; 11:3, s. 193-203
  • Journal article (peer-reviewed)abstract
    • This paper analyzes subband beamforming schemes mainly aimed at speech enhancement and acoustic echo suppression applications such as hands-free telephony for both mobile and office environments, internet telephony and video conferencing. Analytical descriptions of both causal finite-length and noncausal infinite-length subband microphone array structures are given. More specifically, this paper compares finite Wiener filter performance with the noncausal Wiener solution, giving a comprehensive theoretical suppression limit. It is shown that even short filters will yield a good approximation of the infinite solution, provided that the element spacing and temporal sampling is matched to the frequency band of interest. Typically, 10-20 FIR taps are sufficient in each subband.
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3.
  • Schüldt, Christian, et al. (author)
  • Adaptive filter length selection for acoustic echo cancellation
  • 2009
  • In: Signal Processing. - AMSTERDAM : Elsevier. - 0165-1684 .- 1872-7557. ; 89:6, s. 1185-1194
  • Journal article (peer-reviewed)abstract
    • The number of coefficients in an adaptive finite impulse response filter-based acoustic echo cancellation setup is an important parameter, affecting the overall performance of the echo cancellation. Too few coefficients give undermodelling and too many cause slow convergence and an additional echo due to the mismatch of the extra coefficients. This paper proposes a method to adaptively determine the filter length, based on estimation of the mean square deviation. The method is primarily intended for identifying long non-sparse systems, such as a typical impulse response from an acoustic setup. Simulations with band limited flat spectrum signals are used for verification, showing the behavior and benefits of the proposed algorithm. Furthermore, off-line calculation using recorded speech signals show the behavior in real situations and comparison with another state-of-the-art variable filter length algorithm shows the advantages of the proposed method.
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4.
  • Yermeche, Zohra, et al. (author)
  • Bind Subband Beamforming with Time-delay Constraints for Moving Source Speech Enhancement
  • 2007
  • In: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:8, s. 2360-2372
  • Journal article (peer-reviewed)abstract
    • 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.
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5.
  • Berg, Magnus, et al. (author)
  • A combined on-line PCA-ICA algorithm for blind source separation
  • 2005
  • Conference paper (peer-reviewed)abstract
    • In this paper, we introduce and investigate a method combining principle component analysis (PCA) and independent component analysis (ICA) for blind source separation (BSS). A recursive method for the PCA is applied to meet the demands of a real-time application, and likewise a modified on-line version of the information maximization principle is used. The combined PCA-ICA algorithm, first reduces the dimension of the problem and then separates the signals. Evaluation of the proposed algorithm in a real room shows superior noise suppression capabilities compared to the use of PCA or ICA individually. The proposed algorithm achieves an impressive noise suppression/separation of up to 14 dB with only two microphones. Most importantly this is achieved with negligible distortion of the recovered signal. © 2005 IEEE.
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6.
  • Grbic, Nedelko, et al. (author)
  • A New Pilot-Signal based Space-Time Adaptive Algorithm
  • 2001
  • Conference paper (peer-reviewed)abstract
    • In the application of adaptive antenna arrays to wireless communications, a known pilot signal sequence may be used for estimating the array response at the beginning of each data frame. This pilot sequence is usually very short and conventional training methods which estimate the array response, based solely on this training sequence, may incur large estimation errors. In this paper, we propose an online modified weighted recursive least squares type of training algorithm for estimating the optimal array response by exploiting information from the whole frame of the received signal. The benefits of the proposed algorithm is that tracking of coherent noise and interference signals is substantially improved, and the overall performance is increased. Simulation results show that the proposed method offers substantial improvement when compared to the conventional least squares method.
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7.
  • Khan, Muhammad Gufran, et al. (author)
  • A Doublet-Shift Transmitted Reference Scheme for Ultra-Wideband Communication Systems
  • 2007
  • Conference paper (peer-reviewed)abstract
    • The transmitted reference (TR) scheme has gained attention for ultra-wideband (UWB) communications as it bypasses the complex task of channel estimation and provides a simple receiver structure. However, the low-complexity detection in this scheme is achieved at the expense of a loss in performance. To this end, this paper proposes a doublet-shift TR (DsTR) signaling and detection scheme to improve the performance of conventional TR impulse radio UWB system. The performance of the proposed scheme is compared with conventional TR signaling scheme in terms of uncoded bit error rate (BER). The simulation results validate that the doublet-shift TR scheme achieves about 1.5-2 dB better performance than the conventional TR scheme. The DsTR scheme does not require extra energy and still has the low-complexity.
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8.
  • Khan, Muhammad Gufran, et al. (author)
  • Detection of Impulse Radio Ultra-Wideband Signals using Recursive Transmitted Reference Receivers
  • 2007
  • Conference paper (peer-reviewed)abstract
    • For the detection of impulse radio (IR) UWB signals, complex RAKE receiver or low-complexity transmitted reference (TR) autocorrelation receivers can be used. However, there is a performance loss associated with the low-complexity TR scheme. This paper proposes recursive structures of the conventional TR and averaged TR schemes to improve the detection performance of IR-UWB signals. The performance of proposed schemes is evaluated over the standard IEEE 802.15.4a multipath channels. The performance is compared with conventional TR receivers in terms of uncoded bit-error-rate (BER), assuming that the channel is quasi-static. For averaged and recursive averaged TR schemes, the TR sequence is also slightly modified. The simulation results validate that the proposed schemes has better performance by about 2 dB than the conventional TR and averaged TR receivers.
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9.
  • Khan, Muhammad Gufran, et al. (author)
  • Performance Evaluation of RAKE Receiver for UWB Systems using Measured Channels in Industrial Environments
  • 2006
  • Conference paper (peer-reviewed)abstract
    • The industrial environments are an important scenario for ultra wideband (UWB) communication systems. However, due to large number of metallic scatterers in the surroundings, the multipath offered by UWB channels is dense with significant energy. In this paper, the performance of RAKE receivers operating in a non line-of-sight (NLOS) scenario in these environments is evaluated. The channels used for the evaluation are measured in a medium-sized industrial environment. In addition, a standard IEEE 802.15.4a channel model is used for comparison with the results of the measured data. The performance of partial RAKE (PRake) and selective RAKE (SRake)is evaluated in terms of uncoded bit-error-rate (BER) using different number of fingers. The performance of maximal ratio combining (MRC) and equal gain combining (EGC) is compared for the RAKE receiver assuming perfect knowledge of the channel state. Finally, based on the simulation results, conclusions are drawn considering the performance and complexity issues for system design in these environments.
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10.
  • Khan, Muhammad Gufran, et al. (author)
  • Robust Weighted Non-Coherent Receiver for Impulse Radio UWB PPM Signals
  • 2011
  • In: IEEE Communications Letters. - : IEEE Communications Society. - 1089-7798 .- 1558-2558. ; 15:06, s. 614-616
  • Journal article (peer-reviewed)abstract
    • This letter proposes an energy detection based robust weight estimation scheme for pulse-position modulated (PPM) impulse radio ultra-wideband (IR-UWB) signals using weighted non-coherent receiver (WNCR). Conventional data-aided WNCR (DA-WNCR) scheme estimates the weighting coefficients, or channel state information (CSI), using a large number of training symbols over time-varying channels. In contrast, the proposed Robust WNCR (R-WNCR) scheme is non-data-aided (NDA), adaptive and robust to channel variations. The proposed R-WNCR estimates the weighting coefficients adaptively based on the received stochastic data, and the weight estimation process is refined using a decision directed approach.
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  • Result 1-10 of 361
Type of publication
conference paper (243)
journal article (77)
reports (25)
book chapter (11)
doctoral thesis (3)
licentiate thesis (2)
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Type of content
peer-reviewed (315)
other academic/artistic (46)
Author/Editor
Håkansson, Lars (87)
Nordholm, Sven (75)
Dahl, Mattias (65)
Nordebo, Sven (62)
Johansson, Sven (58)
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Lagö, Thomas L (57)
Grbic, Nedelko (41)
Lindström, Fredric (33)
Nordberg, Jörgen (27)
Sällberg, Benny (19)
Schüldt, Christian (18)
Persson, Per (16)
Winberg, Mathias (16)
Nilsson, Mikael (14)
Larsson, Martin (12)
Bengtsson, Bengt (11)
Gustavsson, Ingvar (10)
Mohammed, Abbas (9)
Khan, Muhammad Gufra ... (9)
Andrén, Linus (9)
Yermeche, Zohra (9)
Eriksson, Per (8)
Zackrisson, Johan (8)
Pettersson, Mats (7)
Bartunek, Josef Strö ... (7)
Håkansson, Lars, 196 ... (7)
Nilsson, Kristian (6)
Adolfsson, Stefan (5)
Berggren, Magnus (4)
Brandt, Anders (4)
Bahrami, Ali (4)
Sturesson, Per-Olof (4)
Borgh, Markus, 1983- (4)
Bolmsjö, Gunnar (3)
Lövström, Benny (3)
Cornelius, Per (3)
Tufvesson, Fredrik (2)
Nilsson, Andreas (2)
Ståhl, Jan-Eric (2)
Gertsovich, Irina (2)
Shahid, Muhammad (2)
Blomberg, Mats (2)
Svensson, Tobias (2)
García-Zubía, Javier (2)
Sigfridsson, Andreas (2)
Low, Siow Yong (2)
Waye, Kerstin Persso ... (2)
Claesson, Lena (2)
Khan, Imran, 1978- (2)
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University
Blekinge Institute of Technology (352)
Lund University (13)
Linnaeus University (12)
Luleå University of Technology (2)
Umeå University (1)
Malmö University (1)
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Red Cross University College (1)
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Language
English (355)
Swedish (6)
Research subject (UKÄ/SCB)
Engineering and Technology (361)
Natural sciences (3)
Social Sciences (2)
Medical and Health Sciences (1)

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