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  1. Most audio compression formats are based on the idea of low bit rate transparent encoding. As these types of audio signals are starting to migrate from portable players with inexpensive headphones to higher qu...

    Authors: Demetrios Cantzos, Athanasios Mouchtaris and Chris Kyriakakis

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:462830

    Content type: Research Article

    Published on:

  2. We propose a novel approach to improve adaptive decorrelation filtering- (ADF-) based speech source separation in diffuse noise. The effects of noise on system adaptation and separation outputs are handled sep...

    Authors: Rong Hu and Yunxin Zhao

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:349214

    Content type: Research Article

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  3. This paper proposes a new algorithm for a directional aid with hearing defenders. Users of existing hearing defenders experience distorted information, or in worst cases, directional information may not be per...

    Authors: Benny Sällberg, Farook Sattar and Ingvar Claesson

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:274684

    Content type: Research Article

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  4. We propose a new low complexity, low delay, and fast converging frequency-domain adaptive algorithm for network echo cancellation in VoIP exploiting MMax and sparse partial (SP) tap-selection criteria in the f...

    Authors: Xiang(Shawn) Lin, Andy W.H. Khong, Milŏs Doroslovăcki and Patrick A. Naylor

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:156960

    Content type: Research Article

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  5. Binaural cue coding (BCC) is an efficient technique for spatial audio rendering by using the side information such as interchannel level difference (ICLD), interchannel time difference (ICTD), and interchannel...

    Authors: Bo Qiu, Yong Xu, Yadong Lu and Jun Yang

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:618104

    Content type: Research Article

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  6. The behavior of time delay estimation (TDE) is well understood and therefore attractive to apply in acoustic source localization (ASL). A time delay between microphones maps into a hyperbola. Furthermore, the ...

    Authors: Pasi Pertilä, Teemu Korhonen and Ari Visa

    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2008 2008:278185

    Content type: Research Article

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  7. 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

    Content type: Research Article

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  8. 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

    Content type: Research Article

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  9. 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

    Content type: Research Article

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  10. 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

    Content type: Research Article

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  11. 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

    Content type: Research Article

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  12. 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

    Content type: Research Article

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  13. 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

    Content type: Research Article

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  14. 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

    Content type: Research Article

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  15. 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

    Content type: Research Article

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  16. 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

    Content type: Research Article

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  17. 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

    Content type: Research Article

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  18. 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

    Content type: Research Article

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  19. 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

    Content type: Research Article

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  20. 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

    Content type: Research Article

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  21. 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

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  22. 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

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  23. 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

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  24. 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

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  25. 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

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  26. 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

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  27. 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

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  28. 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

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  29. 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

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  30. 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

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  31. 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

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  32. 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

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  33. 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:

  34. 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

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  35. 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

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  36. 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

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  37. 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:

  38. 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:

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