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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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:071495
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:092528
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:084376
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:096101
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:064506
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:085438
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:078439
We introduce an efficient hidden Markov model-based voice activity detection (VAD) algorithm with time-variant state-transition probabilities in the underlying Markov chain. The transition probabilities vary i...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:043218
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 ...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:034242
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:031314
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:079032
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 ...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:010231
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:084186
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:047891
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...
Citation: EURASIP Journal on Audio, Speech, and Music Processing 2006 2007:039546
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Speed
75 days to first decision for reviewed manuscripts only
64 days to first decision for all manuscripts
222 days from submission to acceptance
34 days from acceptance to publication
Citation Impact
1.558 - 2-year Impact Factor
2.008 - 5-year Impact Factor
1.101 - Source Normalized Impact per Paper (SNIP)
0.259 - SCImago Journal Rank (SJR)
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