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EURASIP Journal on Audio, Speech, and Music Processing, Volume 2014
Volume 2014, 2014
- Changchun Bao, Xin Liu, Yong-tao Sha, Xingtao Zhang:
A blind bandwidth extension method for audio signals based on phase space reconstruction. 1 - Madoka Miki, Norihide Kitaoka, Chiyomi Miyajima, Takanori Nishino, Kazuya Takeda:
Improvement of multimodal gesture and speech recognition performance using time intervals between gestures and accompanying speech. 2 - Jeongook Song, Hong-Goo Kang:
New aliasing cancelation algorithm for the transition between non-aliased and TDAC-based coding modes. 3 - Xie Sun, Yunxin Zhao:
Integrated exemplar-based template matching and statistical modeling for continuous speech recognition. 4 - Ryo Aihara, Ryoichi Takashima, Tetsuya Takiguchi, Yasuo Ariki:
A preliminary demonstration of exemplar-based voice conversion for articulation disorders using an individuality-preserving dictionary. 5 - Daniel Pape
, Luis M. T. Jesus
:
Production and perception of velar stop (de)voicing in European Portuguese and Italian. 6 - Gen-Fang Chen
, Jia-Shing Sheu
:
An optical music recognition system for traditional Chinese Kunqu Opera scores written in Gong-Che Notation. 7 - Mohammadali Sebghati
, Hamidreza Amindavar
, James A. Ritcey:
Basis approach to estimate the instantaneous frequencies in multicomponent AM-FM signals. 8 - Toni Heittola, Annamaria Mesaros
, Dani Korpi
, Antti J. Eronen, Tuomas Virtanen
:
Method for creating location-specific audio textures. 9 - Shi Dong
, Ruimin Hu, Xiaochen Wang, Yuhong Yang, Weiping Tu:
Expanded three-channel mid/side coding for three-dimensional multichannel audio systems. 10 - Wen-Lin Zhang, Wei-Qiang Zhang, Dan Qu, Bi-Cheng Li:
Speaker adaptation based on regularized speaker-dependent eigenphone matrix estimation. 11 - Soheil Khorram, Hossein Sameti, Fahimeh Bahmaninezhad, Simon King
, Thomas Drugman:
Context-dependent acoustic modeling based on hidden maximum entropy model for statistical parametric speech synthesis. 12 - Aditya Arie Nugraha
, Kazumasa Yamamoto, Seiichi Nakagawa:
Single-channel dereverberation by feature mapping using cascade neural networks for robust distant speaker identification and speech recognition. 13 - Ján Stas
, Jozef Juhár
, Daniel Hládek
:
Classification of heterogeneous text data for robust domain-specific language modeling. 14 - Zhaofeng Zhang, Longbiao Wang, Atsuhiko Kai:
Distant-talking speaker identification by generalized spectral subtraction-based dereverberation and its efficient computation. 15 - Jack Leegaard, Jan Østergaard
, Søren Holdt Jensen, Ram Zamir:
Practical design of delta-sigma multiple description audio coding. 16 - Yue Xian Zou, Peng Wang, Yongqing Wang, Christian H. Ritz
, Jiangtao Xi
:
Speech enhancement with an acoustic vector sensor: an effective adaptive beamforming and post-filtering approach. 17 - Rupayan Chakraborty
, Climent Nadeu, Taras Butko:
Source ambiguity resolution of overlapped sounds in a multi-microphone room environment. 18 - Yongzhe Shi, Wei-Qiang Zhang, Meng Cai, Jia Liu:
Empirically combining unnormalized NNLM and back-off N-gram for fast N-best rescoring in speech recognition. 19 - Upal Mahbub, Shaikh Anowarul Fattah, Wei-Ping Zhu
, M. Omair Ahmad:
Single-channel acoustic echo cancellation in noise based on gradient-based adaptive filtering. 20 - Hari Krishna Maganti, Marco Matassoni:
Auditory processing-based features for improving speech recognition in adverse acoustic conditions. 21 - Sumxin Jiang, Rendong Ying, Peilin Liu:
A Memory Efficient Finite State Source Coding Algorithm for Audio MDCT Coefficients. 22 - Santitham Prom-on, Peter Birkholz, Yi Xu:
Identifying underlying articulatory targets of Thai vowels from acoustic data based on an analysis-by-synthesis approach. 23 - Yi-Ju Lin
, Tien-Ming Wang, Ta-Chun Chen, Yin-Lin Chen, Wei-Chen Chang, Alvin Wen-Yu Su:
Musical note analysis of solo violin recordings using recursive regularization. 25 - Francisco J. Cañadas-Quesada
, Pedro Vera-Candeas
, Nicolás Ruiz-Reyes
, Julio J. Carabias-Orti
, Pablo Cabañas Molero
:
Percussive/harmonic sound separation by non-negative matrix factorization with smoothness/sparseness constraints. 26 - Rafael P. Ribeiro, Murilo Almeida, Carlos Nascimento Silla Jr.
:
The Ethnic Lyrics Fetcher tool. 27 - Humberto M. Torres
, Jorge A. Gurlekian, Hansjörg Mixdorff, Hartmut R. Pfitzinger:
Linguistically motivated parameter estimation methods for a superpositional intonation model. 28 - Ye Jiang, Kong-Aik Lee
, Longbiao Wang:
PLDA in the I-Supervector Space for Text-Independent Speaker Verification. 29 - Ning-Yun Ku, Cheng-Yu Yeh, Shaw-Hwa Hwang:
An efficient algebraic codebook search for ACELP speech coder. 30 - Stephan Gerlach, Jörg Bitzer
, Stefan Goetze
, Simon Doclo
:
Joint estimation of pitch and direction of arrival: improving robustness and accuracy for multi-speaker scenarios. 31 - Yanna Ma, Akinori Nishihara:
A modified Wiener filtering method combined with wavelet thresholding multitaper spectrum for speech enhancement. 32 - Hansjörg Mixdorff, Adrian Leemann
, Volker Dellwo
:
The influence of speech rate on Fujisaki model parameters. 33 - Diego Castán, Alfonso Ortega
, Antonio Miguel, Eduardo Lleida
:
Audio segmentation-by-classification approach based on factor analysis in broadcast news domain. 34 - Devireddy Hanumantha Rao Naidu, Sriram Srinivasan:
Robust Bayesian estimation for context-based speech enhancement. 35 - Sara Ahmadi, Seyed Ahadi, Bert Cranen, Lou Boves:
Sparse coding of the modulation spectrum for noise-robust automatic speech recognition. 36 - Ali Al-Haj:
An imperceptible and robust audio watermarking algorithm. 37 - Gilles Degottex
, Daniel Erro:
A uniform phase representation for the harmonic model in speech synthesis applications. 38 - Marek Bohac, Michaela Kucharová, Zoraida Callejas
, Jan Nouza, Petr Cerva
:
A cross-lingual adaptation approach for rapid development of speech recognizers for learning disabled users. 39 - Furong Yan, Yanbin Zhang, Jiachang Yan:
A sub-band-based feature reconstruction approach for robust speaker recognition. 40 - Xin Liu, Changchun Bao:
Audio bandwidth extension based on temporal smoothing cepstral coefficients. 41 - Wei-Wei Liu, Wei-Qiang Zhang, Michael T. Johnson, Jia Liu:
Homogenous ensemble phonotactic language recognition based on SVM supervector reconstruction. 42 - Bart Ons, Jort F. Gemmeke, Hugo Van hamme
:
The self-taught vocal interface. 43
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