Skip to main content

Articles

Page 11 of 11

  1. 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
  2. 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
  3. 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
  4. 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
  5. This paper proposes an audio-visual speech recognition method using lip information extracted from side-face images as an attempt to increase noise robustness in mobile environments. Our proposed method assume...

    Authors: Koji Iwano, Tomoaki Yoshinaga, Satoshi Tamura and Sadaoki Furui
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:064506
  6. 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
  7. 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
  8. 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
  9. 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
  10. 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
  11. 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
  12. 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
  13. 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
  14. 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

Who reads the journal?

Learn more about the impact the EURASIP Journal on Audio, Speech, and Music Processing has worldwide

Annual Journal Metrics

  • 2022 Citation Impact
    2.4 - 2-year Impact Factor
    2.0 - 5-year Impact Factor
    1.081 - SNIP (Source Normalized Impact per Paper)
    0.458 - SJR (SCImago Journal Rank)

    2022 Speed
    16 days submission to first editorial decision for all manuscripts (Median)
    185 days submission to accept (Median)

    2022 Usage 
    249,941 downloads
    65 Altmetric mentions 

Funding your APC

​​​​​​​Open access funding and policy support by SpringerOpen​​

​​​​We offer a free open access support service to make it easier for you to discover and apply for article-processing charge (APC) funding. Learn more here