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  1. Rhythmic information plays an important role in Music Information Retrieval. Example applications include automatically annotating large databases by genre, meter, ballroom dance style or tempo, fully automate...

    Authors: Björn Schuller, Florian Eyben and Gerhard Rigoll
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:846135
  2. The phasor representation is introduced to identify the characteristic of the active noise control (ANC) systems. The conventional representation, transfer function, cannot explain the fact that the performanc...

    Authors: Fu-Kun Chen, Ding-Horng Chen and Yue-Dar Jou
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:126859
  3. A multiresolution source/filter model for coding of audio source signals (spot recordings) is proposed. Spot recordings are a subset of the multimicrophone recordings of a music performance, before the mixing ...

    Authors: Athanasios Mouchtaris, Kiki Karadimou and Panagiotis Tsakalides
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:624321
  4. The automatic recognition of foreign-accented Arabic speech is a challenging task since it involves a large number of nonnative accents. As well, the nonnative speech data available for training are generally ...

    Authors: YousefAjami Alotaibi, Sid-Ahmed Selouani and Douglas O'Shaughnessy
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:679831
  5. This paper deals with continuous-time filter transfer functions that resemble tuning curves at particular set of places on the basilar membrane of the biological cochlea and that are suitable for practical VLS...

    Authors: AG Katsiamis, EM Drakakis and RF Lyon
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:063685
  6. This work is the result of an interdisciplinary collaboration between scientists from the fields of audio signal processing, phonetics and cognitive neuroscience aiming at studying the perception of modificati...

    Authors: Sølvi Ystad, Cyrille Magne, Snorre Farner, Gregory Pallone, Mitsuko Aramaki, Mireille Besson and Richard Kronland-Martinet
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:030194
  7. Multistage vector quantization (MSVQ) is a technique for low complexity implementation of high-dimensional quantizers, which has found applications within speech, audio, and image coding. In this paper, a mult...

    Authors: Pradeepa Yahampath
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:067146
  8. Variability of speaker accent is a challenge for effective human communication as well as speech technology including automatic speech recognition and accent identification. The motivation of this study is to ...

    Authors: Ayako Ikeno and John HL Hansen
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:076030
  9. A noise suppression algorithm is proposed based on filtering the spectrotemporal modulations of noisy signals. The modulations are estimated from a multiscale representation of the signal spectrogram generated...

    Authors: Nima Mesgarani and Shihab Shamma
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:042357
  10. We describe two voice-to-phoneme conversion algorithms for speaker-independent voice-tag creation specifically targeted at applications on embedded platforms. These algorithms (batch mode and sequential) are comp...

    Authors: YanMing Cheng, Changxue Ma and Lynette Melnar
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2008:568737
  11. This paper experimentally shows the importance of perceptual continuity of the expressive strength in vocal timbre for natural change in vocal expression. In order to synthesize various and continuous expressi...

    Authors: Tomoko Yonezawa, Noriko Suzuki, Shinji Abe, Kenji Mase and Kiyoshi Kogure
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:023807
  12. Many modern speech bandwidth extension techniques predict the high-frequency band based on features extracted from the lower band. While this method works for certain types of speech, problems arise when the c...

    Authors: Visar Berisha and Andreas Spanias
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:016816
  13. Wide band digital audio signals have a very high data-rate associated with them due to their complex nature and demand for high-quality reproduction. Although recent technological advancements have significant...

    Authors: Karthikeyan Umapathy and Sridhar Krishnan
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:051563
  14. This paper proposes a new technique for improving the performance of linear prediction analysis by utilizing a refined version of the autocorrelation function. Problems in analyzing voiced speech using linear ...

    Authors: M Shahidur Rahman and Tetsuya Shimamura
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2007 2007:045962
  15. 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
  16. 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
  17. 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
  18. 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
  19. 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
  20. 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
  21. 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
  22. 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
  23. 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
  24. 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
  25. 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
  26. 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
  27. 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
  28. 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
  29. 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
  30. 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
  31. 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
  32. 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

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