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  1. We propose a novel approach of integrating exemplar-based template matching with statistical modeling to improve continuous speech recognition. We choose the template unit to be context-dependent phone segment...

    Authors: Xie Sun and Yunxin Zhao
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2014 2014:4
  2. This paper proposes a new aliasing cancelation algorithm for the transition between non-aliased coding and transform coding with time domain aliasing cancelation (TDAC). It is effectively utilized for unified ...

    Authors: Jeongook Song and Hong-Goo Kang
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2014 2014:3
  3. We propose an integrative method of recognizing gestures such as pointing, accompanying speech. Speech generated simultaneously with gestures can assist in the recognition of gestures, and since this occurs in...

    Authors: Madoka Miki, Norihide Kitaoka, Chiyomi Miyajima, Takanori Nishino and Kazuya Takeda
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2014 2014:2
  4. Bandwidth extension is an effective technique for enhancing the quality of audio signals by reconstructing their high-frequency components. In this paper, a novel blind bandwidth extension method is proposed b...

    Authors: Chang-Chun Bao, Xin Liu, Yong-Tao Sha and Xing-Tao Zhang
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2014 2014:1
  5. Prosody and prosodic boundaries carry significant information regarding linguistics and paralinguistics and are important aspects of speech. In the field of prosodic event detection, many local acoustic featur...

    Authors: Junhong Zhao, Wei-Qiang Zhang, Hua Yuan, Michael T Johnson, Jia Liu and Shanhong Xia
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:30
  6. In this paper, we propose a novel noise-robustness method known as weighted sub-band histogram equalization (WS-HEQ) to improve speech recognition accuracy in noise-corrupted environments. Considering the obse...

    Authors: Jeih-weih Hung and Hao-teng Fan
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:29
  7. The framework of voice conversion system is expected to emphasize both the static and dynamic characteristics of the speech signal. The conventional approaches like Mel frequency cepstrum coefficients and line...

    Authors: Jagannath H Nirmal, Mukesh A Zaveri, Suprava Patnaik and Pramod H Kachare
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:28
  8. This paper investigates real-time N-dimensional wideband sound source localization in outdoor (far-field) and low-degree reverberation cases, using a simple N-microphone arrangement. Outdoor sound source localiza...

    Authors: Ali Parsayan and Seyed Mohammad Ahadi
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:27

    The Correction to this article has been published in EURASIP Journal on Audio, Speech, and Music Processing 2022 2022:24

  9. Affective computing, especially from speech, is one of the key steps toward building more natural and effective human-machine interaction. In recent years, several emotional speech corpora in different languag...

    Authors: Caglar Oflazoglu and Serdar Yildirim
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:26
  10. The performance of thresholding-based methods for speech enhancement largely depends upon the estimation of the exact threshold value. In this paper, a new thresholding-based speech enhancement approach, where...

    Authors: Tahsina Farah Sanam and Celia Shahnaz
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:25
  11. This paper investigates multi-modal aspects of audiovisual quality assessment for interactive communication services. It shows how perceived auditory and visual qualities integrate to an overall audiovisual qu...

    Authors: Benjamin Belmudez and Sebastian Möller
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:24
  12. Query-by-Example Spoken Term Detection (QbE STD) aims at retrieving data from a speech data repository given an acoustic query containing the term of interest as input. Nowadays, it has been receiving much int...

    Authors: Javier Tejedor, Doroteo T Toledano, Xavier Anguera, Amparo Varona, Lluís F Hurtado, Antonio Miguel and José Colás
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:23
  13. The recurrent neural network language model (RNNLM) has shown significant promise for statistical language modeling. In this work, a new class-based output layer method is introduced to further improve the RNN...

    Authors: Yongzhe Shi, Wei-Qiang Zhang, Jia Liu and Michael T Johnson
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:22
  14. This paper proposes a novel and robust voice activity detection (VAD) algorithm utilizing long-term spectral flatness measure (LSFM) which is capable of working at 10 dB and lower signal-to-noise ratios(SNRs)....

    Authors: Yanna Ma and Akinori Nishihara
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:87

    The Erratum to this article has been published in EURASIP Journal on Audio, Speech, and Music Processing 2015 2015:30

  15. Many features have been proposed for speech-based emotion recognition, and a majority of them are frame based or statistics estimated from frame-based features. Temporal information is typically modelled on a ...

    Authors: Vidhyasaharan Sethu, Eliathamby Ambikairajah and Julien Epps
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:19
  16. Nonnegative matrix factorization (NMF) is developed for parts-based representation of nonnegative signals with the sparseness constraint. The signals are adequately represented by a set of basis vectors and th...

    Authors: Jen-Tzung Chien and Hsin-Lung Hsieh
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:18
  17. In this study, we focus on the classification of neutral and stressed speech based on a physical model. In order to represent the characteristics of the vocal folds and vocal tract during the process of speech...

    Authors: Xiao Yao, Takatoshi Jitsuhiro, Chiyomi Miyajima, Norihide Kitaoka and Kazuya Takeda
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:17
  18. This paper presents a bimodal acoustic-visual synthesis technique that concurrently generates the acoustic speech signal and a 3D animation of the speaker’s outer face. This is done by concatenating bimodal di...

    Authors: Slim Ouni, Vincent Colotte, Utpala Musti, Asterios Toutios, Brigitte Wrobel-Dautcourt, Marie-Odile Berger and Caroline Lavecchia
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:16
  19. Most existing automatic chord recognition systems use a chromagram in front-end processing and some sort of classifier (e.g., hidden Markov model, Gaussian mixture model (GMM), support vector machine, or other...

    Authors: Maksim Khadkevich and Maurizio Omologo
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:15
  20. Cochannel speech separation aims to separate two speech signals from a single mixture. In a supervised scenario, the identities of two speakers are given, and current methods use pre-trained speaker models for...

    Authors: Ke Hu and DeLiang Wang
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:14
  21. A challenging open question in music classification is which music representation (i.e., audio features) and which machine learning algorithm is appropriate for a specific music classification task. To address...

    Authors: Yannis Panagakis and Constantine Kotropoulos
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:13
  22. Multiple-model based speech recognition (MMSR) has been shown to be quite successful in noisy speech recognition. Since it employs multiple hidden Markov model (HMM) sets that correspond to various noise types...

    Authors: Yongjoo Chung and John HL Hansen
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:12
  23. A novel speech bandwidth extension method based on audio watermark is presented in this paper. The time-domain and frequency-domain envelope parameters are extracted from the high-frequency components of speec...

    Authors: Zhe Chen, Chengyong Zhao, Guosheng Geng and Fuliang Yin
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:10
  24. This paper presents a novel lossless compression technique of the context-based adaptive arithmetic coding which can be used to further compress the quantized parameters in audio codec. The key feature of the ...

    Authors: Jing Wang, Xuan Ji, Shenghui Zhao, Xiang Xie and Jingming Kuang
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:9
  25. This article analyzes and compares influence of different types of spectral and prosodic features for Czech and Slovak emotional speech classification based on Gaussian mixture models (GMM). Influence of initi...

    Authors: Jiří Přibil and Anna Přibilová
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:8
  26. In this article, we describe a speaker adaptation method based on the probabilistic 2-mode analysis of training models. Probabilistic 2-mode analysis is a probabilistic extension of multilinear analysis. We ap...

    Authors: Yongwon Jeong
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:7
  27. Availability of large amounts of raw unlabeled data has sparked the recent surge in semi-supervised learning research. In most works, however, it is assumed that labeled and unlabeled data come from the same d...

    Authors: Konstantin Markov and Tomoko Matsui
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:6
  28. A comprehensive system for facial animation of generic 3D head models driven by speech is presented in this article. In the training stage, audio-visual information is extracted from audio-visual training data...

    Authors: Lucas D Terissi, Mauricio Cerda, Juan C Gómez, Nancy Hitschfeld-Kahler and Bernard Girau
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:5
  29. Blind source separation (BSS) and sound activity detection (SAD) from a sound source mixture with minimum prior information are two major requirements for computational auditory scene analysis that recognizes ...

    Authors: Kohei Nagira, Takuma Otsuka and Hiroshi G Okuno
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:4
  30. We propose an efficient solution to the problem of sparse linear prediction analysis of the speech signal. Our method is based on minimization of a weighted l2-norm of the prediction error. The weighting function...

    Authors: Vahid Khanagha and Khalid Daoudi
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:3
  31. A lot of effort has been made in Computational Auditory Scene Analysis (CASA) to segregate target speech from monaural mixtures. Based on the principle of CASA, this article proposes an improved algorithm for ...

    Authors: Wang Yu, Lin Jiajun, Chen Ning and Yuan Wenhao
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:2
  32. The work presented in this article studies how the context information can be used in the automatic sound event detection process, and how the detection system can benefit from such information. Humans are usi...

    Authors: Toni Heittola, Annamaria Mesaros, Antti Eronen and Tuomas Virtanen
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2013 2013:1
  33. This article describes a modified technique for enhancing noisy speech to improve automatic speech recognition (ASR) performance. The proposed approach improves the widely used spectral subtraction which inher...

    Authors: Hari Krishna Maganti and Marco Matassoni
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:29
  34. Conventional parametric stereo (PS) audio coding employs inter-channel phase difference and overall phase difference as phase parameters. In this article, it is shown that those parameters cannot correctly rep...

    Authors: Dong-il Hyun, Young-cheol Park and Dae Hee Youn
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:27
  35. In this article, the authors propose an optimally designed fixed beamformer (BF) for stereophonic acoustic echo cancelation (SAEC) in real hands-free communication applications. Several contributions related t...

    Authors: Matteo Pirro, Stefano Squartini, Laura Romoli and Francesco Piazza
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:26
  36. The rapid spread in digital data usage in many real life applications have urged new and effective ways to ensure their security. Efficient secrecy can be achieved, at least in part, by implementing steganogra...

    Authors: Fatiha Djebbar, Beghdad Ayad, Karim Abed Meraim and Habib Hamam
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:25
  37. Mood is an important aspect of music and knowledge of mood can be used as a basic feature in music recommender and retrieval systems. A listening experiment was carried out establishing ratings for various moo...

    Authors: Bert den Brinker, Ralph van Dinther and Janto Skowronek
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:24
  38. A vast amount of audio features have been proposed in the literature to characterize the content of audio signals. In order to overcome specific problems related to the existing features (such as lack of discrimi...

    Authors: Toni Mäkinen, Serkan Kiranyaz, Jenni Raitoharju and Moncef Gabbouj
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:23
  39. Humans exhibit a remarkable ability to reliably classify sound sources in the environment even in presence of high levels of noise. In contrast, most engineering systems suffer a drastic drop in performance wh...

    Authors: Sridhar Krishna Nemala, Dmitry N Zotkin, Ramani Duraiswami and Mounya Elhilali
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:22
  40. A new method to secure speech communication using the discrete wavelet transforms (DWT) and the fast Fourier transform is presented in this article. In the first phase of the hiding technique, we separate the ...

    Authors: Siwar Rekik, Driss Guerchi, Sid-Ahmed Selouani and Habib Hamam
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:20
  41. In this article, we present the evaluation results for the task of speaker diarization of broadcast news, which was part of the Albayzin 2010 evaluation campaign of language and speech technologies. The evalua...

    Authors: Martin Zelenák, Henrik Schulz and Javier Hernando
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:19
  42. Dance movements are a complex class of human behavior which convey forms of non-verbal and subjective communication that are performed as cultural vocabularies in all human cultures. The singularity of dance f...

    Authors: João Lobato Oliveira, Luiz Naveda, Fabien Gouyon, Luis Paulo Reis, Paulo Sousa and Marc Leman
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:18
  43. In this article, we propose a new set of acoustic features for automatic emotion recognition from audio. The features are based on the perceptual quality metrics that are given in perceptual evaluation of audi...

    Authors: Mehmet Cenk Sezgin, Bilge Gunsel and Gunes Karabulut Kurt
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:16
  44. An increasing number of multilingual applications require language recognition (LRE) as a frontend, but desire low additional computational cost. This article demonstrates a novel architecture for embedding ph...

    Authors: Yuxiang Shan, Yan Deng, Jia Liu and Michael T Johnson
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:15
  45. The problem of blind source separation (BSS) of convolved acoustic signals is of great interest for many classes of applications. Due to the convolutive mixing process, the source separation is performed in th...

    Authors: Eugen Hoffmann, Dorothea Kolossa, Bert-Uwe Köhler and Reinhold Orglmeister
    Citation: EURASIP Journal on Audio, Speech, and Music Processing 2012 2012:14

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