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  1. 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

    Content type: Research Article

    Published on:

  2. 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

    Content type: Research Article

    Published on:

  3. 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

    Content type: Research Article

    Published on:

  4. 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

    Content type: Research Article

    Published on:

  5. This paper investigates the significance of combining cepstral features derived from the modified group delay function and from the short-time spectral magnitude like the MFCC. The conventional group delay fun...

    Authors: Rajesh M. Hegde, Hema A. Murthy and V. R. R. Gadde

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:079032

    Content type: Research Article

    Published on:

  6. 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

    Content type: Research Article

    Published on:

  7. We present a new technique for separating two speech signals from a single recording. The proposed method bridges the gap between underdetermined blind source separation techniques and those techniques that model...

    Authors: Mohammad H. Radfar, Richard M. Dansereau and Abolghasem Sayadiyan

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:084186

    Content type: Research Article

    Published on:

  8. Animated agents are becoming increasingly frequent in research and applications in speech science. An important challenge is to evaluate the effectiveness of the agent in terms of the intelligibility of its vi...

    Authors: Slim Ouni, Michael M Cohen, Hope Ishak and Dominic W Massaro

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:047891

    Content type: Research Article

    Published on:

  9. This paper presents a new method to detect speech/nonspeech components of a given noisy signal. Employing the combination of binary Walsh basis functions and an analysis-synthesis scheme, the original noisy sp...

    Authors: Moe Pwint and Farook Sattar

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:039546

    Content type: Research Article

    Published on:

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