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  1. 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
  2. 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
  3. 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
  4. 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
  5. 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
  6. 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
  7. 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
  8. 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
  9. 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
  10. 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
  11. 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
  12. 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
  13. 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
  14. 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
  15. 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
  16. 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
  17. 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
  18. 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
  19. 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
  20. 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
  21. 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
  22. 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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