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  1. A novel approach for robust dialogue act detection in a spoken dialogue system is proposed. Shallow representation named partial sentence trees are employed to represent automatic speech recognition outputs. P...

    Authors: Chia-Ping Chen, Chung-Hsien Wu and Wei-Bin Liang
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:13
  2. As fundamental research for human-robot interaction, this paper addresses the rhythmic reference of a human while turning a rope with another human. We hypothyzed that when interpreting rhythm cues to make a r...

    Authors: Kenta Yonekura, Chyon Hae Kim, Kazuhiro Nakadai, Hiroshi Tsujino and Shigeki Sugano
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:12
  3. This article proposes a new acoustic model using decision trees (DTs) as replacements for Gaussian mixture models (GMM) to compute the observation likelihoods for a given hidden Markov model state in a speech ...

    Authors: Masami Akamine and Jitendra Ajmera
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:10
  4. A study on force-feedback interaction with a model of a neural oscillator provides insight into enhanced human-robot interactions for controlling musical sound. We provide differential equations and discrete-t...

    Authors: Edgar Berdahl, Claude Cadoz and Nicolas Castagné
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:9
  5. Interaction with human musicians is a challenging task for robots as it involves online perception and precise synchronization. In this paper, we present a consistent and theoretically sound framework for comb...

    Authors: Umut Şimşekli, Orhan Sönmez, Barış Kurt Kurt and Ali Taylan Cemgil
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:8
  6. The aim of this paper is to improve beat-tracking for live guitar performances. Beat-tracking is a function to estimate musical measurements, for example musical tempo and phase. This method is critical to ach...

    Authors: Tatsuhiko Itohara, Takuma Otsuka, Takeshi Mizumoto, Angelica Lim, Tetsuya Ogata and Hiroshi G Okuno
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:6
  7. This study proposes a music-aided framework for affective interaction of service robots with humans. The framework consists of three systems, respectively, for perception, memory, and expression on the basis o...

    Authors: Jeong-Sik Park, Gil-Jin Jang and Yong-Ho Seo
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:5
  8. This article studies a vital issue in wireless communications, which is the transmission of audio signals over wireless networks. It presents a novel interleaver scheme for protection against error bursts and ...

    Authors: Mohsen Ahmed Mahmoud Mohamed El-Bendary, Atef E Abou-El-azm, Nawal A El-Fishawy, Farid Shawki, Fathi E Abd-ElSamie, Mostafa Ali Refai El-Tokhy and Hassan B Kazemian
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:4
  9. It has been long speculated that expression of emotions from different modalities have the same underlying 'code', whether it be a dance step, musical phrase, or tone of voice. This is the first attempt to imp...

    Authors: Angelica Lim, Tetsuya Ogata and Hiroshi G Okuno
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:3
  10. The availability of haptic interfaces in music content processing offers interesting possibilities of performer-instrument interaction for musical expression. These new musical instruments can precisely modulate ...

    Authors: Victor Zappi, Antonio Pistillo, Sylvain Calinon, Andrea Brogni and Darwin Caldwell
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:2
  11. Most of voice activity detection (VAD) schemes are operated in the discrete Fourier transform (DFT) domain by classifying each sound frame into speech or noise based on the DFT coefficients. These coefficients...

    Authors: Shiwen Deng and Jiqing Han
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:12
  12. The main objective of the work presented in this paper was to develop a complete system that would accomplish the original visions of the MALACH project. Those goals were to employ automatic speech recognition...

    Authors: Josef Psutka, Jan Švec, Josef V Psutka, Jan Vaněk, Aleš Pražák, Luboš Šmídl and Pavel Ircing
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:10
  13. In this article, a novel technique based on the empirical mode decomposition methodology for processing speech features is proposed and investigated. The empirical mode decomposition generalizes the Fourier an...

    Authors: Kuo-Hau Wu, Chia-Ping Chen and Bing-Feng Yeh
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:9
  14. This article proposes a multiscale product (MP)-based method for estimating the open quotient (OQ) from the speech waveform. The MP is operated by calculating the wavelet transform coefficients of the speech s...

    Authors: Wafa Saidi, Aicha Bouzid and Noureddine Ellouze
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:8
  15. In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ra...

    Authors: Markus Borgh, Magnus Berggren, Christian Schüldt, Fredric Lindström and Ingvar Claesson
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:7
  16. The first large vocabulary speech recognition system for the Persian language is introduced in this paper. This continuous speech recognition system uses most standard and state-of-the-art speech and language ...

    Authors: Hossein Sameti, Hadi Veisi, Mohammad Bahrani, Bagher Babaali and Khosro Hosseinzadeh
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:6
  17. The spectrum subtraction method is one of the most common methods by which to remove noise from a spectrum. Like many noise reduction methods, the spectrum subtraction method uses discrete Fourier transform (D...

    Authors: Toshio Yoshizawa, Shigeki Hirobayashi and Tadanobu Misawa
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:5
  18. This work studies the task of automatic emotion detection in music. Music may evoke more than one different emotion at the same time. Single-label classification and regression cannot model this multiplicity. ...

    Authors: Konstantinos Trohidis, Grigorios Tsoumakas, George Kalliris and Ioannis Vlahavas
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:4
  19. The frequency-to-channel mapping for Cochlear implant (CI) signal processors was originally designed to optimize speech perception and generally does not preserve the harmonic structure of music sounds. An alg...

    Authors: Sherif Abdellatif Omran, Waikong Lai, Michael Büchler and Norbert Dillier
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:2
  20. Recently, audio segmentation has attracted research interest because of its usefulness in several applications like audio indexing and retrieval, subtitling, monitoring of acoustic scenes, etc. Moreover, a pre...

    Authors: Taras Butko and Climent Nadeu
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:1
  21. Authors: Bhiksha Raj, Paris Smaragdis, Malcolm Slaney, Chung-Hsien Wu, Liming Chen and Hyoung-Gook Kim
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2010:467278
  22. We address the question of whether and how boosting and bagging can be used for speech recognition. In order to do this, we compare two different boosting schemes, one at the phoneme level and one at the utter...

    Authors: Christos Dimitrakakis and Samy Bengio
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2011:426792
  23. To overcome harmonic structure distortions of complex tones in the low frequency range due to the frequency to electrode mapping function used in Nucleus cochlear implants, two modified frequency maps based on...

    Authors: Sherif A. Omran, Waikong Lai and Norbert Dillier
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2011 2010:948565
  24. This paper describes a novel approach for localization of multiple sources overlapping in time. The proposed algorithm relies on acoustic maps computed in multi-microphone settings, which are descriptions of t...

    Authors: Alessio Brutti, Maurizio Omologo and Piergiorgio Svaizer
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:147495
  25. Correlogram is an important representation for periodic signals. It is widely used in pitch estimation and source separation. For these applications, major problems of correlogram are its low resolution and re...

    Authors: Xueliang Zhang, Wenju Liu and Bo Xu
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:252374
  26. In this paper we present a method to search for environmental sounds in large unstructured databases of user-submitted audio, using a general sound events taxonomy from ecological acoustics. We discuss the use...

    Authors: Gerard Roma, Jordi Janer, Stefan Kersten, Mattia Schirosa, Perfecto Herrera and Xavier Serra
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:960863
  27. Degrouping is the key component in MPEG Layer II audio decoding. It mainly contains the arithmetic operations of division and modulo. So far no dedicated degrouping algorithm and architecture is well realized....

    Authors: Tsung-Han Tsai
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:737450
  28. Organizing a database of user-contributed environmental sound recordings allows sound files to be linked not only by the semantic tags and labels applied to them, but also to other sounds with similar acoustic...

    Authors: Gordon Wichern, Brandon Mechtley, Alex Fink, Harvey Thornburg and Andreas Spanias
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:192363
  29. A multimicrophone speech enhancement algorithm for binaural hearing aids that preserves interaural time delays was proposed recently. The algorithm is based on multichannel Wiener filtering and relies on a voi...

    Authors: Jasmina Catic, Torsten Dau, JörgM Buchholz and Fredrik Gran
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:840294
  30. A method is described for quantifying the quality of wideband speech codecs. Two parameters are derived from signal-based speech quality model estimations: (i) a wideband equipment impairment factor

    Authors: Sebastian Möller, Nicolas Côté, Valérie Gautier-Turbin, Nobuhiko Kitawaki and Akira Takahashi
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:782731
  31. In multiway loudspeaker systems, digital signal processing techniques have been used to correct the frequency response, the propagation time, and the lobbing errors. These solutions are mainly based on correct...

    Authors: Hmaied Shaiek and JeanMarc Boucher
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:928439
  32. Humans represent sounds to others and receive information about sounds from others using onomatopoeia. Such representation is useful for obtaining and reporting the acoustic features and impressions of actual ...

    Authors: Masayuki Takada, Nozomu Fujisawa, Fumino Obata and Shin-ichiro Iwamiya
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:674248
  33. We give a brief discussion on the amplitude and frequency variation rates of the sinusoid representation of signals. In particular, we derive three inequalities that show that these rates are upper bounded by ...

    Authors: Xue Wen and Mark Sandler
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:941732
  34. This paper presents a method for estimating the amplitude of coincident partials generated by harmonic musical sources (instruments and vocals). It was developed as an alternative to the commonly used interpol...

    Authors: JaymeGarciaArnal Barbedo and George Tzanetakis
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:523791
  35. Nowadays, audio podcasting has been widely used by many online sites such as newspapers, web portals, journals, and so forth, to deliver audio content to users through download or subscription. Within 1 to 30 ...

    Authors: MN Nguyen, Qi Tian and Ping Xue
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:572571
  36. Frequency-domain blind source separation (BSS) performs poorly in high reverberation because the independence assumption collapses at each frequency bins when the number of bins increases. To improve the separ...

    Authors: Lin Wang, Heping Ding and Fuliang Yin
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:797962
  37. Speaker identification performance is almost perfect in neutral talking environments. However, the performance is deteriorated significantly in shouted talking environments. This work is devoted to proposing, ...

    Authors: Ismail Shahin
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:862138
  38. Theoretical and applied environmental sounds research is gaining prominence but progress has been hampered by the lack of a comprehensive, high quality, accessible database of environmental sounds. An ongoing ...

    Authors: Brian Gygi and Valeriy Shafiro
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:654914
  39. This paper presents a model-based method for coding the LSF parameters of LPC speech coders on a "long-term" basis, that is, beyond the usual 20–30 ms frame duration. The objective is to provide efficient LSF ...

    Authors: Laurent Girin
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:597039
  40. Authors: Georg Stemmer, Elmar Nöth and Vijay Parsa
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:835974
  41. When a number of speakers are simultaneously active, for example in meetings or noisy public places, the sources of interest need to be separated from interfering speakers and from each other in order to be ro...

    Authors: Dorothea Kolossa, Ramon Fernandez Astudillo, Eugen Hoffmann and Reinhold Orglmeister
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:651420
  42. The aim of the study is to transpose and extend to a set of environmental sounds the notion of sound descriptors usually used for musical sounds. Four separate primary studies dealing with interior car sounds,...

    Authors: Nicolas Misdariis, Antoine Minard, Patrick Susini, Guillaume Lemaitre, Stephen McAdams and Etienne Parizet
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2010 2010:362013

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