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ORCIDEUROSPEECH 1995: Madrid, Spain
- Fourth European Conference on Speech Communication and Technology, EUROSPEECH 1995, Madrid, Spain, September 18-21, 1995. ISCA 1995
Keynotes
- Kenneth N. Stevens:
Applying phonetic knowledge to lexical access. 3-14 - William A. Ainsworth:
Auditory mechanisms for speech perception. 171-178 - Hervé Bourlard:
Towards increasing speech recognition error rates. 883-894 - Sadaoki Furui:
Flexible speech recognition. 1595-1604
Prosody Modelling in ASR
- Takatoshi Jitsuhiro, Tomokazu Yamada, Shigeki Sagayama:
Syllabic duration control for vocabulary-free speech recognition. 15-18 - Kazuyuki Takagi, Shuichi Itahashi:
Effectiveness of pause information in the content word detection of spoken dialogues. 19-22 - Kazuhiro Kondo:
Connected Japanese digit recognition with pitch accent-dependent models. 23-26 - Claude Barras, Marie-José Caraty, Claude Montacié:
Temporal control and training selection for HMM-based system. 27-30 - Keikichi Hirose, Xinhui Hu:
HMM-based tone recognition of Chinese trisyllables using double codebooks on fundamental frequency and waveform power. 31-34
Wideband Coding
- C. Murgia, Gang Feng, Catherine Quinquis, Alain Le Guyader:
Very low delay and high quality coding of 20 hz -15 khz speech at 64 kbit/S. 37-40 - Shigeaki Sasaki, Akitoshi Kataoka, Takehiro Moriya:
Wideband CELP coder at 16-kbit/s with 10-ms frame. 41-44 - A. W. Black, Ian A. Atkinson, Ahmet M. Kondoz, Barry G. Evans:
High quality 14.1kb/s wideband speech coder. 45-48 - Victoria Abreu-Sernández, Domingo Docampo-Amoedo:
A multipulse-deconvolution codec for wideband speech. 49-52 - Thierry Chonavel, Samir Saoudi:
Multi-channel linear predictive coding of audio signals. 53-56
Hardware and Systems for Speech Processing
- E. Rohwer:
An advanced multi-DSP platform for speech technology integration in computer telephony applications. 58-62 - Rafael Ciria, Rafael Sarmiento de Sotomayor, Cristina Aguila, José Parera, Juan Santos:
Voice processing architecture for computer-telephony integration. 63-66 - Laura Ortiz-Balbuena, Héctor M. Pérez Meana, Alejandro Martínez-González, Luís Nino de Rivera, Mariko Nakano-Miyatake:
Fast convergent analog adaptive filter. 67-70 - Manuel A. Leandro, Álvaro Villegas, José Manuel Pardo:
Efficient isolated word recognition in Spanish based on static modeling. 71-74 - Jean-Luc Cochard, Olivier Oppizzi:
Reliability in a multi-agent spoken language recognition system. 75-78
Discriminative Training I, II
- Tomoko Matsui, Sadaoki Furui:
A study of speaker adaptation based on minimum classification error training. 81-84 - Albino Nogueiras Rodríguez, José B. Mariño:
Maximum likelihood based discriminative training of acoustic models. 85-88 - Hugues Leprieur, Patrick Haffner:
Discriminant learning with minimum memory loss for improved non-vocabulary rejection. 89-92 - Cesar Martín del Alamo, F. Javier Caminero-Gil, Celinda de la Torre-Munilla, Luis A. Hernández Gómez:
Codebook weights adaptation for discriminative training of SCHMM-based speech recognition systems. 93-96 - Kyungmin Na, Bumki Jeon, Dong-Il Chang, Soo-Ik Chae, Souguil Ann:
Discriminative training of hidden Markov models using overall risk criterion and reduced gradient method. 97-100 - Qiang Huo, Chorkin Chan:
Discriminative training of HMM based speech recognizer with gradient projection method. 101-104 - Javier Hernando Pericas, J. Ayarte, Enric Monte:
Optimization of speech parameter weighting for CDHMM word recognition. 105-108 - Mazin G. Rahim, Chin-Hui Lee, Biing-Hwang Juang:
Discriminative utterance verification for connected digits recognition. 529-532 - Antonio M. Peinado, Antonio J. Rubio, José C. Segura, Victoria E. Sánchez, Jesús Esteban Díaz Verdejo:
MCE estimation of VQ parameters for MVQHMM speech recognition. 533-536 - Wolfgang Reichl, Günther Ruske:
Discriminative training for continuous speech recognition. 537-540 - Kuldip K. Paliwal, Michiel Bacchiani, Yoshinori Sagisaka:
Minimum classification error training algorithm for feature extractor and pattern classifier in speech recognition. 541-544
Auditory Modeling in Speech Recognition
- Stephan Euler:
Integrated optimization of feature transformation for speech recognition. 109-112 - Andrew C. Morris, José M. Pardo:
Phoneme transition detection and broad classification using a simple model based on the function of onset detector cells found in the cochlear nucleus. 115-118 - Eric Fragnière, André van Schaik, Eric A. Vittoz:
Linear predictive coding of speech using an analogue cochlear model. 119-122 - Edward Jones, Eliathamby Ambikairajah:
Pitch extraction of telephone bandwidth speech using a place-temporal approach. 123-126 - Markus Bodden, Timothy R. Anderson:
A binaural selectivity model for speech recognition. 127-130 - Cristina Dobrin, Petri Haavisto, Kari Laurila, Jaakko Astola:
Speech recognition experiments in a noisy environment using auditory system modelling. 131-134
Applications and Systems
- Frédéric Berthommier, Georg F. Meyer:
Source separation by a functional model of amplitude demodulation. 135-138 - Vratislav Davidek, Pavel Sovka, Jiri Sika:
Real-time implementation of spectral subtraction algorithm for suppression of acoustic noise in speech. 141-144 - Bert Van Coile, Hans-Wilhelm Rühl, L. Vogten, M. Thoone, S. Goß, D. Delaey, E. Moons, Jacques M. B. Terken, Jan-Roelof de Pijper, Marianne Kugler, P. Kaufholz, Regina Krüger, Steven Leys, S. Willems:
Speech synthesis for the new pan-european traffic message control system RDS-TMC. 145-148 - Roberto Pacifici, G. Manca:
Echo cancelling in speech recognition systems. 149-152 - Teodoro Calonge Cano, Luis Alonso, Rui Ralha, A. L. Sánchez:
Parallel implementation of an hybrid neural network used for speech recognition task. 153-156 - M. Li, J. T. Proudfoot:
Hardware design of LPC coding for speech feature extraction. 157-160 - Henning Bergmann, Hans-Hermann Hamer, Andreas Noll, Annedore Paeseler, Horst Tomaschewski:
Modularization in task-specific language modelling. 161-164 - Carlos Avendaño, Hynek Hermansky, Eric A. Wan:
Beyond NYQUIST: towards the recovery of broad-bandwidth speech from narrow-bandwidth speech. 165-168
Large Vocabulary
- David Pye, Philip C. Woodland, Steve J. Young:
Large vocabulary multilingual speech recognition using HTK. 181-184 - Lori Lamel, Martine Adda-Decker, Jean-Luc Gauvain:
Issues in Large Vocabulary, Multilingual Speech Recognition. 185-188 - James Barnett, Paul G. Bamberg, Martin Held, Juan Huerta, Linda Manganaro, Adam Weiss:
Comparative performance in large-vocabulary isolated-word recognition in five european languages. 189-192 - Julie Brousseau, Caroline Drouin, George F. Foster, Pierre Isabelle, Roland Kuhn, Yves Normandin, Pierre Plamondon:
French speech recognition in an automatic dictation system for translators: the transtalk project. 193-196 - Christian Dugast, Xavier L. Aubert, Reinhard Kneser:
The Philips large-vocabulary recognition system for american English, French, and German. 197-200 - Sung-Chien Lin, Lee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee:
A syllable-based very-large-vocabulary voice retrieval system for Chinese databases with textual attributes. 203-206 - Michael Riley, Andrej Ljolje, Donald Hindle, Fernando Pereira:
The AT&t 60,000 word speech-to-text system. 207-210 - Tai-Hsuan Ho, Hsin-Min Wang, Lee-Feng Chien, Keh-Jiann Chen, Lin-Shan Lee:
Fast and accurate continuous speech recognition for Chinese language with very large vocabulary. 211-214 - Zuoying Wang, Jun Wu, Xi Xiao, Jin Quo:
Methods towards the very large vocabulary Chinese speech recognition. 215-218 - Gary D. Cook, Anthony J. Robinson:
Utterance clustering for large vocabulary continuous speech recognition. 219-222
Speech Coding I
- Thomas Eriksson, Jan Linden, Jan Skoglund:
Vector quantization of glottal pulses. 225-228 - Michele Festa, Daniele Sereno:
A speech coding algorithm based on prototypes interpolation with critical bands and phase coding. 229-232 - Dionysis E. Tsoukalas, Jiannis Mouropoulos, George Kokkinakis:
Very low-bitrate speech coding using perceptually-derived spectral data. 233-236 - Lorenzo Piazzo:
A new very low bit rate speech coder: the step decomposition vocoder. 237-240 - Ian A. Atkinson, Ahmet M. Kondoz, Barry G. Evans:
Time envelope LP vocoder: a new coding technique at very low bit rates. 241-244 - Dan Stefanoiu, Radwan Kastantin, Gang Feng:
Speech coding based on the discrete-time wavelet transform and human auditory system properties. 661-664 - F. J. Ancin, M. L. Larreategui, B. L. Burrows, Rolando A. Carrasco:
Wavelets for low bit rate speech coding applications. 665-669 - Elimberaza Mandridake, Rachid Atay, Mohamed Najim:
Adaptive speech vector coding with a multiresolution hierarchical codebook. 669-672 - Andrei Popescu, Nicolas Moreau:
Subband analysis-by-synthesis coding. 673-676 - Clifford I. Parris, Danny Wong, Francois Chambon:
A robust 2.4kb/s LP-MBE with iterative LP modelling. 677-680 - M. S. Torres-Guijarro, Francisco Javier Casajús-Quirós:
Improved transient representation and quantization for sinusoidal speech coders. 681-684 - Eric W. M. Yu, Cheung-Fat Chan:
Efficient multiband excitation linear predictive coding of speech at 1.6 kbps. 685-688 - Bruno Wery, Stephane Deketelaere:
Voice coding in the MSBN satellite communication system. 689-692 - Barry M. G. Cheetham, Xiaoqin Sun, W. T. K. Wong:
Spectral envelope estimation for low bit-rate sinusoidal speech coders. 693-696
Speech Signal Processing / Wavelets
- Israel Cohen, Shalom Raz, David Malah:
Shift-invariant adaptive local trigonometric decomposition. 247-250 - Paul Micallef, Edward H. S. Chilton:
Spectral envelope of speech using wavelets. 251-254 - Andrzej Drygajlo, Nicolas Thevoz:
Multiresolution speech analysis using fast time-varying orthogonal wavelet packet transform algorithms. 255-258 - Maria Rangoussi, Flemming Pedersen:
Second- and third-order wigner distributions in hierarchical recognition of speech phonemes. 259-262 - Gaafar M. K. Saleh, Mahesan Niranjan, William J. Fitzgerald:
The use of maximum a posteriori parameters in linear prediction of speech. 263-268
Applications of Speech Technology
- William C. G. Ortel:
Observed long-term changes in customer calling patterns in a telephone application using automatic speech recognition. 269-272 - Ayman Asadi, David M. Lubensky, L. Madhavrao, Jayant M. Naik, Vijay Raman, George Vysotsky:
Combining speech algorithms into a "natural" application of speech technology for telephone network services. 273-276 - Yevgeny Ludovik, Valeriy Sibirtsev:
Intelligent answering machine-secretary. 277-280 - Kyung-ho Loken-Kim, Young-Duk Park, Suguru Mizunashi, Laurel Fais, Tsuyoshi Morimoto:
Verbal-gestural behaviors in multimodal spoken language interpreting telecommunications. 281-284 - Jung-Kuei Chen, Lin-Shan Lee, Frank K. Soong:
Large vocabulary, word-based Mandarin dictation system. 285-288
Visual Speech
- Bertrand Le Goff, Thierry Guiard-Marigny, Christian Benoît:
Read my lips... and my jaw! how intelligible are the components of a speaker's face? 291-294 - Angela Fuster Duran:
Mcgurk effect in Spanish and German listeners: influences of visual cues in the perception of Spanish and German conflicting audio-visual stimuli. 295-298 - Jonas Beskow:
Rule-based visual speech synthesis. 299-302 - Fabio Lavagetto, Paolo Lavagetto:
A new algorithm for visual synthesis of speech. 303-306 - Harouna Kabré:
Audiovisual speech recognition using the fuzzy shape filters model. 307-310
Speaker Recognition I-III
- Jialong He, Li Liu, Günther Palm:
On the use of features from prediction residual signals in speaker identification. 313-316 - Kai Tat Ng, Haizhou Li, Jean Paul Haton:
Some nonparametric distance measures in speaker verification. 317-320 - Michael J. Carey, Graham Tattersall, Eluned S. Parris:
Adaptive transforms for speaker recognition. 321-324 - Kai Tat Ng, Jian Su, Bingzheng Xu:
Speaker recognition with discriminative speaker VQ models. 325-328 - A. Federico, Andrea Paoloni:
Parametric speaker recognition over large population of telephonic voices. 329-332 - Toomas Altosaar, Einar Meister:
Speaker recognition experiments in Estonian using multi-layer feed-forward neural nets. 333-336 - Ivan Magrin-Chagnolleau, Jean-François Bonastre, Frédéric Bimbot:
Effect of utterance duration and phonetic content on speaker identification using second-order statistical methods. 337-340 - Pavel V. Labulin, Sergey L. Koval, Andrey N. Raev:
Automatic speaker recognition using formants-based nearest-neighbour distance measure. 341-344 - Mohammad Mehdi Homayounpour, Gérard Chollet:
Discrimination of voices of twins and siblings for speaker verification. 345-348 - Kay M. Berkling, Etienne Barnard:
Theoretical error prediction for a language identification system using optimal phoneme clustering. 351-354 - Jesper Ø. Olsen:
Separation of speakers in audio data. 355-358 - J.-L. Bonifas, I. Hernaez Rioja, B. Etxebarria Gonzalez, S. Saoudi:
Text-dependent speaker verification using dynamic time warping and vector quantization of LSF. 359-362 - Haizhou Li, Jean Paul Haton, Yifan Gong:
On MMI learning of Gaussian mixture for speaker models. 363-366 - Yifan Gong:
Evaluation of Bayes decision approach to automatic determination of thresholds for speaker verification. 367-370 - Daniele Falavigna:
Comparison of different HMM based methods for speaker verification. 371-374 - J. Sheikhzadegan, M. Tebiani, M. Lotfizad, Mahmood R. Roohani:
Speaker classification by neural network for short utteranses using phoneme groups in Farsi. 375-378 - Jean-Luc Le Floch, Claude Montacié, Marie-José Caraty:
Speaker recognition experiments on the NTIMIT database. 379-382 - Michael Wagner, John S. Mason, J. Bruce Millar:
Speaker identification using vector quantisation with codeword-specific derivative coding. 383-386 - Haizhou Li, Jean Paul Haton, Jian Su, Yifan Gong:
Speaker recognition with temporal transition models. 617-620 - Tomoko Matsui, Tomohito Kanno, Sadaoki Furui:
Speaker recognition using HMM composition in noisy environments. 621-624 - ChiWei Che, Qiguang Lin:
Speaker recognition using HMM with experiments on the yoho database. 625-628 - Kin Yu, John S. Mason, John Oglesby:
Speaker recognition models. 629-632 - Thierry Artières, Patrick Gallinari:
Multi-state predictive neural networks for text-independent speaker recognition. 633-636
Voice Source Analysis and Modelling
- Francesco Beritelli, Salvatore Casale, Marco Russo:
A voiced/unvoiced speech discrimination technique based on fuzzy logic. 389-392 - Vassilios Darsinos, Christophe d'Alessandro, B. Yegnanarayana:
Evaluation of a periodic/aperiodic speech decomposition algorithm. 393-396 - Jean Rouat, Yong Chun Liu, Daniel Morissette:
A pitch determination and voiced/unvoiced decision algorithm for noisy speech. 397-400 - Léonard Janer:
Modulated Gaussian wavelet transform based speech analyser (MGWTSA) pitch detection algorithm (PDA). 401-404 - Juan L. Navarro-Mesa, Ignasi Esquerra-Llucia:
A time-frequency approach to epoch detection. 405-408 - M. L. Larreategui, F. J. Ancin, Rolando A. Carrasco:
An improved epoch detection algorithm based on sinusoidal modelling of speech. 409-412 - Vassilios Darsinos, Dimitrios Galanis, George Kokkinakis:
A method for fully automatic analysis and modelling of voice source characteristics. 413-416 - Hartmut R. Pfitzinger:
Dynamic vowel quality: a new determination formalism based on perceptual experiments. 417-420 - Sumio Ohno, Hiroya Fujisaki:
A method for quantitative analysis of the local speech rate. 421-424
Voice Personality Characteristics in TTS
- Ki-Seung Lee, Dae Hee Youn, Il-Whan Cha:
Voice personality transformation using an orthogonal vector space conversion. 427-430 - Makoto Hashimoto, Norio Higuchi:
Spectral mapping for voice conversion using speaker selection and vector field smoothing. 431-435 - Norio Higuchi, Makoto Hashimoto:
Analysis of acoustic features affecting speaker identification. 435-438 - Masato Akagi, Taw Ienaga:
Speaker individualities in fundamental frequency contours and its control. 439-442 - King-fai Lam, Cheung-Fat Chan:
Interpolating MBE v/UV mixture function for high quality synthesis of speech. 443-447 - Yannis Stylianou, Olivier Cappé, Eric Moulines:
Statistical methods for voice quality transformation. 447-450 - Yannis Stylianou, Jean Laroche, Eric Moulines:
High-quality speech modification based on a harmonic + noise model. 451-454