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Adaptive Partial-Update and Sparse System Identification

  1. Content type: Research Article

    In various adaptive estimation applications, such as acoustic echo cancellation within teleconferencing systems, the input signal is a highly correlated speech. This, in general, leads to extremely slow conver...

    Authors: Yan Wu Jennifer, John Homer, Geert Rombouts and Marc Moonen

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:071495

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  3. Content type: Research Article

    We investigate novel algorithms to improve the convergence and reduce the complexity of time-domain convolutive blind source separation (BSS) algorithms. First, we propose MMax partial update time-domain convo...

    Authors: Qiongfeng Pan and Tyseer Aboulnasr

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:092528

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  4. Content type: Research Article

    A sparse system identification algorithm for network echo cancellation is presented. This new approach exploits both the fast convergence of the improved proportionate normalized least mean square (IPNLMS) alg...

    Authors: Andy W.H. Khong, Patrick A. Naylor and Jacob Benesty

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:084376

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  5. Content type: Research Article

    The μ-law proportionate normalized least mean square (MPNLMS) algorithm has been proposed recently to solve the slow convergence problem of the proportionate normalized least mean square (PNLMS) algorithm afte...

    Authors: Hongyang Deng and Miloš Doroslovački

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:096101

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  6. Content type: Research Article

    This paper introduces new algorithms for the blind separation of audio sources using modal decomposition. Indeed, audio signals and, in particular, musical signals can be well approximated by a sum of damped s...

    Authors: Abdeldjalil Aïssa-El-Bey, Karim Abed-Meraim and Yves Grenier

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:085438

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  7. Content type: Research Article

    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 syst...

    Authors: Fredric Lindstrom, Christian Schüldt and Ingvar Claesson

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:078439

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  8. Content type: Research Article

    Proportionate adaptive filters can improve the convergence speed for the identification of sparse systems as compared to their conventional counterparts. In this paper, the idea of proportionate adaptation is ...

    Authors: Stefan Werner, José A Apolinário Jr. and Paulo S R Diniz

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:034242

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  9. Content type: Research Article

    The paper provides an analysis of the transient and the steady-state behavior of a filtered-x partial-error affine projection algorithm suitable for multichannel active noise control. The analysis relies on energ...

    Authors: Alberto Carini and Giovanni L Sicuranza

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:031314

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  10. Content type: Research Article

    This paper derives an upper bound for the step size of the sequential partial update (PU) LMS adaptive algorithm when the input signal is a periodic reference consisting of several harmonics. The maximum step ...

    Authors: Pedro Ramos, Roberto Torrubia, Ana López, Ana Salinas and Enrique Masgrau

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:010231

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