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Sökning: WFRF:(Ingvar Christian) > Blekinge Tekniska Högskola

  • Resultat 1-10 av 19
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1.
  • Berggren, Magnus, et al. (författare)
  • Low-complexity network echo cancellation approach for systems equipped with external memory
  • 2011
  • Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 19:8, s. 2506-2515
  • Tidskriftsartikel (refereegranskat)abstract
    • Long delays and sparseness characterize impulse responses in telecommunication networks and a vast number of solutions for network echo cancellation have been proposed over the years. In this paper, an approach for detecting dispersive regions of a sparse impulse response and a proportionate normalized least mean square (PNLMS)-based selective updating approach are combined with an adaptive double-talk detector to form a complete solution for echo cancellation. The proposed solution has low computational complexity and is targeted for systems equipped with external memory.
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2.
  • Borgh, Markus, 1983-, et al. (författare)
  • An improved adaptive gain equalizer for noise reduction with low speech distortion
  • 2011
  • Ingår i: EURASIP Journal on Audio, Speech, and Music Processing. - : Springer. - 1687-4714 .- 1687-4722. ; 7
  • Tidskriftsartikel (refereegranskat)abstract
    • In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.
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4.
  • Garcia-Loro, Felix, et al. (författare)
  • Experimenting in PILAR federation : A common path for the future
  • 2018
  • Ingår i: PROCEEDINGS OF 2018 IEEE GLOBAL ENGINEERING EDUCATION CONFERENCE (EDUCON) - EMERGING TRENDS AND CHALLENGES OF ENGINEERING EDUCATION. - : IEEE Computer Society. - 9781538629574 ; , s. 1518-1523
  • Konferensbidrag (refereegranskat)abstract
    • The PILAR (Platform Integration of Laboratories based on the Architecture of visiR) Erasmus Plus project started in September 2016 and will last three years. The core of the PILAR project is the VISIR remote laboratory-Virtual Instruments System In Reality-. The project aims for a federation of five of the existing VISIR nodes, sharing experiments, capacity and resources among partners, and to provide access to VISIR remote lab, through PILAR consortium, to students from other educational institutions. PILAR will be the framework from which management tasks will be performed and laboratories/experiments will be shared. PILAR will also foster the Special Interest Group of VISIR under the Global Online Laboratory Consortium (GOLC) of the International Association of Online Engineering (IAOE). © 2018 IEEE.
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5.
  • Lindström, Fredric, et al. (författare)
  • A hybrid acoustic echo canceller and suppressor
  • 2007
  • Ingår i: Signal Processing. - : Elsevier. - 0165-1684. ; 87:4, s. 739-749
  • Tidskriftsartikel (refereegranskat)abstract
    • Wideband communication is becoming a desired feature in telephone conferencing systems. This paper proposes a computationally efficient echo suppression control algorithm to be used when increasing the bandwidth of an audio conferencing system, e.g. a conference telephone. The method presented in this paper gives a quality improvement, in the form of increased bandwidth, at a negligible extra computational cost. The increase in bandwidth is obtained through combining a conventional acoustic echo cancellation unit and an acoustic echo suppression unit, i.e. a hybrid echo canceller and suppressor. The proposed solution was implemented in a real-time system. Frequency analysis combined with subjective tests showed that the proposed method extends the bandwidth, while maintaining high quality.
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6.
  • Lindström, Fredric, et al. (författare)
  • A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation
  • 2007
  • Ingår i: IEEE transactions on circuits and systems I-Regular Papers. - : IEEE. - 1057-7122. ; 54:9, s. 2011-2018
  • Tidskriftsartikel (refereegranskat)abstract
    • The two-path algorithm is an adaptive filter algorithm based on a parallel filter structure, which has been found to be useful for line echo cancellation as well as for acoustic echo cancellation. It is well known that in finite precision arithmetic, the adaptation process of adaptive algorithms can be reduced or even halted due to finite precision effects. This paper proposes a variant of the two-path scheme where the effects of quantization are reduced, without any significant increase in complexity. The improvement is shown by simulations using bandlimited flat spectrum noise as well as real speech signals.
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7.
  • Lindström, Fredric, et al. (författare)
  • An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation
  • 2007
  • Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:4, s. 1320-1326
  • Tidskriftsartikel (refereegranskat)abstract
    • Adaptive filters for echo cancellation generally need update control schemes to avoid divergence in case of significant disturbances. The two-path algorithm avoids the problem of unnecessary halting of the adaptive filter when the control scheme gives an erroneous output. Versions of this algorithm have previously been presented for echo cancellation. This paper presents a transfer logic which improves the convergence speed of the two-path algorithm for acoustic echo cancellation, while retaining the robustness. Results from simulations show an improved performance, and a fixed-point DSP implementation verifies the performance in real-time
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8.
  • Lindström, Fredric, et al. (författare)
  • Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation
  • 2007
  • Ingår i: EURASIP Journal on Audio, Speech, and Music Processing. - : Hindawi Publishing Corporation. - 1687-4714 .- 1687-4722. ; 2007
  • Tidskriftsartikel (refereegranskat)abstract
    • An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter) for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS)-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.
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10.
  • Lindström, Fredric, et al. (författare)
  • Reusing Data During Speech Pauses in an NLMS-based Acoustic Echo Canceller
  • 2006
  • Konferensbidrag (refereegranskat)abstract
    • Fast convergence of the adaptive filter in an acoustic echo cancellation based hands-free communication system is desirable as it implies more periods of possible full-duplex communication. This paper presents a normalized least mean square (NLMS)-based algorithm, targeted for acoustic echo cancellation based units equipped with large external memory. The proposed algorithm utilizes unused processing resources in periods of silence, thus no extra complexity as compared with the conventional NLMS algorithm is required. The improvements obtained by the proposed algorithm are verified through simulated, as well as through real acoustic systems.
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  • Resultat 1-10 av 19

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