Skip to main content

Articles

Page 8 of 8

  1. Recent research on the TIMIT corpus suggests that longer-length acoustic models are more appropriate for pronunciation variation modelling than the context-dependent phones that conventional automatic speech r...

    Authors: Annika Hämäläinen, Lou Boves, Johan de Veth and Louis ten Bosch

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:046460

    Content type: Research Article

    Published on:

  2. When applying automatic speech recognition (ASR) to meeting recordings including spontaneous speech, the performance of ASR is greatly reduced by the overlap of speech events. In this paper, a method of separa...

    Authors: Futoshi Asano, Kiyoshi Yamamoto, Jun Ogata, Miichi Yamada and Masami Nakamura

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:027616

    Content type: Research Article

    Published on:

  3. We describe an FFT-based companding algorithm for preprocessing speech before recognition. The algorithm mimics tone-to-tone suppression and masking in the auditory system to improve automatic speech recogniti...

    Authors: Bhiksha Raj, Lorenzo Turicchia, Bent Schmidt-Nielsen and Rahul Sarpeshkar

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:065420

    Content type: Research Article

    Published on:

  4. Dereverberation is required in various speech processing applications such as handsfree telephony and voice-controlled systems, especially when signals are applied that are recorded in a moderately or highly r...

    Authors: Koen Eneman and Marc Moonen

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:051831

    Content type: Research Article

    Published on:

  5. 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

    Content type: Research Article

    Published on:

  6. 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

    Content type: Research Article

    Published on:

  7. 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

    Content type: Research Article

    Published on:

  8. 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

    Content type: Research Article

    Published on:

  9. 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

    Content type: Research Article

    Published on:

  10. 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:

  11. 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:

  12. 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:

  13. 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:

  14. 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:

  15. 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:

  16. 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:

  17. 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:

  18. 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:

Latest Tweets

Your browser needs to have JavaScript enabled to view this timeline

Who reads the journal?

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

Annual Journal Metrics

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