Shigeki Sagayama

Affiliations:
  • Meiji University, Tokyo, Japan


According to our database1, Shigeki Sagayama authored at least 217 papers between 1986 and 2022.

Collaborative distances:
  • Dijkstra number2 of four.
  • Erdős number3 of four.

Timeline

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Bibliography

2022
Use of Nods Less Synchronized with Turn-Taking and Prosody During Conversations in Adults with Autism.
Proceedings of the Interspeech 2022, 2022

Entrainment Analysis for Assessment of Autistic Speech Prosody Using Bottleneck Features of Deep Neural Network.
Proceedings of the IEEE International Conference on Acoustics, 2022

2021
Semi-automatic music piece creation based on impression words extracted from object and background in color image.
Proceedings of the 10th IEEE Global Conference on Consumer Electronics, 2021

Pitch and Volume Stability in the Communicative Response of Adults with Autism.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2021

2020
A Parameterized Harmony Model for Automatic Music Completion.
J. Inf. Process., 2020

DNN-Based Full-Band Speech Synthesis Using GMM Approximation of Spectral Envelope.
IEICE Trans. Inf. Syst., 2020

Music Recreation in Nursing Home using Automatic Music Accompaniment System and Score of VLN.
Proceedings of the 2nd IEEE Global Conference on Life Sciences and Technologies, 2020

2019
Automatic Music Completion Based on Joint Optimization of Harmony Progression and Voicing.
J. Inf. Process., 2019

Autism Spectrum Disorder Discrimination Based on Voice Activities Related to Fillers and Laughter.
Proceedings of the 53rd Annual Conference on Information Sciences and Systems, 2019

Automatic Piano Reduction of Orchestral Music Based on Musical Entropy.
Proceedings of the 53rd Annual Conference on Information Sciences and Systems, 2019

Piano Practice Evaluation and Visualization by HMM for Arbitrary Jumps and Mistakes.
Proceedings of the 53rd Annual Conference on Information Sciences and Systems, 2019

Polyphonic Voicing Optimization for Automatic Music Completion.
Proceedings of the 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2019

2018
Harmony and Voicing Interpolation for Automatic Music Composition Assistance.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2018

Semi-Supervised NMF in the chroma Domain Applied to Music Harmony Estimation.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2018

Composite Wavelet Model for Stability-Oriented Speech Synthesis from Cepstral Features.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2018

Multiresolutional Hierarchical Bayesian NMF for Detailed Audio Analysis of Music Performances.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2018

2017
Rhythm Transcription of Polyphonic Piano Music Based on Merged-Output HMM for Multiple Voices.
IEEE ACM Trans. Audio Speech Lang. Process., 2017

Variant of Viterbi algorithm based on p-Norm.
Proceedings of the 22nd International Conference on Digital Signal Processing, 2017

Music chord recognition from audio data using bidirectional encoder-decoder LSTMs.
Proceedings of the 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2017

2016
Real-Time Audio-to-Score Alignment of Music Performances Containing Errors and Arbitrary Repeats and Skips.
IEEE ACM Trans. Audio Speech Lang. Process., 2016

A Real-time Audio-to-audio Karaoke Generation System for Monaural Recordings Based on Singing Voice Suppression and Key Conversion Techniques.
J. Inf. Process., 2016

Non-filter waveform generation from cepstrum using spectral phase reconstruction.
Proceedings of the 9th ISCA Speech Synthesis Workshop, 2016

Minimax Viterbi Algorithm for HMM-Based Guitar Fingering Decision.
Proceedings of the 17th International Society for Music Information Retrieval Conference, 2016

2015
Blind Suppression of Nonstationary Diffuse Acoustic Noise Based on Spatial Covariance Matrix Decomposition.
J. Signal Process. Syst., 2015

Autoregressive Hidden Semi-Markov Model of Symbolic Music Performance for Score Following.
Proceedings of the 16th International Society for Music Information Retrieval Conference, 2015

Automatic Piano Reduction from Ensemble Scores Based on Merged-Output Hidden Markov Model.
Proceedings of the Looking Back, 2015

2014
Singing Voice Enhancement in Monaural Music Signals Based on Two-stage Harmonic/Percussive Sound Separation on Multiple Resolution Spectrograms.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

Harmonic/percussive sound separation based on anisotropic smoothness of spectrograms.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

A Stochastic Temporal Model of Polyphonic MIDI Performance with Ornaments.
CoRR, 2014

Outer-Product Hidden Markov Model and Polyphonic MIDI Score Following.
CoRR, 2014

Merged-Output HMM for Piano Fingering of Both Hands.
Proceedings of the 15th International Society for Music Information Retrieval Conference, 2014

Merged-Output Hidden Markov Model for Score Following of MIDI Performance with Ornaments, Desynchronized Voices, Repeats and Skips.
Proceedings of the Music Technology meets Philosophy, 2014

HMM-Based Automatic Arrangement for Guitars with Transposition and its Implementation.
Proceedings of the Music Technology meets Philosophy, 2014

An auxiliary-function approach to online independent vector analysis for real-time blind source separation.
Proceedings of the 4th Joint Workshop on Hands-free Speech Communication and Microphone Arrays, 2014

2013
Dynamic Bayesian Networks for Symbolic Polyphonic Pitch Modeling.
IEEE Trans. Speech Audio Process., 2013

Input-Output HMM Applied to Automatic Arrangement for Guitars.
J. Inf. Process., 2013

Bayesian Nonparametric Approach to Blind Separation of Infinitely Many Sparse Sources.
IEICE Trans. Fundam. Electron. Commun. Comput. Sci., 2013

Text-to-speech synthesizer based on combination of composite wavelet and hidden Markov models.
Proceedings of the Eighth ISCA Tutorial and Research Workshop on Speech Synthesis, 2013

General algorithms for estimating spectrogram and transfer functions of target signal for blind suppression of diffuse noise.
Proceedings of the IEEE International Workshop on Machine Learning for Signal Processing, 2013

Generative modeling of speech F<sub>0</sub> contours.
Proceedings of the INTERSPEECH 2013, 2013

Probabilistic speech F<sub>0</sub> contour model incorporating statistical vocabulary model of phrase-accent command sequence.
Proceedings of the INTERSPEECH 2013, 2013

Probabilistic model of two-dimensional rhythm tree structure representation for automatic transcription of polyphonic MIDI signals.
Proceedings of the Asia-Pacific Signal and Information Processing Association Annual Summit and Conference, 2013

Statistical Approach to Automatic Expressive Rendition of Polyphonic Piano Music.
Proceedings of the Guide to Computing for Expressive Music Performance, 2013

2012
Introduction to the Special Section on Deep Learning for Speech and Language Processing.
IEEE Trans. Speech Audio Process., 2012

Blind Separation of Infinitely Many Sparse Sources.
Proceedings of the IWAENC 2012 - International Workshop on Acoustic Signal Enhancement, Proceedings, RWTH Aachen University, Germany, September 4th, 2012

Context-free 2D Tree Structure Model of Musical Notes for Bayesian Modeling of Polyphonic Spectrograms.
Proceedings of the 13th International Society for Music Information Retrieval Conference, 2012

Variable-length coding of ACELP gain using Entropy-Constrained VQ.
Proceedings of the International Symposium on Communications and Information Technologies, 2012

Hidden Markov Convolutive Mixture Model for Pitch Contour Analysis of Speech.
Proceedings of the INTERSPEECH 2012, 2012

Speaker-Dependent Voice Activity Detection Robust to Background Speech Noise.
Proceedings of the INTERSPEECH 2012, 2012

Assistance for Novice Users on Creating Songs from Japanese Lyrics.
Proceedings of the Non-Cochlear Sound: Proceedings of the 38th International Computer Music Conference, 2012

Comparative evaluations of various harmonic/percussive sound separation algorithms based on anisotropic continuity of spectrogram.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

User-guided independent vector analysis with source activity tuning.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Explicit beat structure modeling for non-negative matrix factorization-based multipitch analysis.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Constrained and regularized variants of non-negative matrix factorization incorporating music-specific constraints.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

A tandem connectionist model using combination of multi-scale spectro-temporal features for acoustic event detection.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

2011
Beyond Timbral Statistics: Improving Music Classification Using Percussive Patterns and Bass Lines.
IEEE ACM Trans. Audio Speech Lang. Process., 2011

Diffuse Noise Suppression Using Crystal-Shaped Microphone Arrays.
IEEE Trans. Speech Audio Process., 2011

Computational auditory induction as a missing-data model-fitting problem with Bregman divergence.
Speech Commun., 2011

Polyphonic Pitch Estimation and Instrument Identification by Joint Modeling of Sustained and Attack Sounds.
IEEE J. Sel. Top. Signal Process., 2011

Introduction to the Special Issue on Music Signal Processing.
IEEE J. Sel. Top. Signal Process., 2011

Bayesian nonparametric spectrogram modeling based on infinite factorial infinite hidden Markov model.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2011

Polyhymnia: An Automatic Piano Performance System with Statistical Modeling of Polyphonic Expression and Musical Symbol Interpretation.
Proceedings of the 11th International Conference on New Interfaces for Musical Expression, 2011

Using Spectral Fluctuation of Speech in Multi-Feature HMM-Based Voice Activity Detection.
Proceedings of the INTERSPEECH 2011, 2011

Concurrent Optimization of Context Clustering and GMM for Offline Handwritten Word Recognition Using HMM.
Proceedings of the 2011 International Conference on Document Analysis and Recognition, 2011

Multipitch estimation by joint modeling of harmonic and transient sounds.
Proceedings of the IEEE International Conference on Acoustics, 2011

Infinite-state spectrum model for music signal analysis.
Proceedings of the IEEE International Conference on Acoustics, 2011

Automatic video annotation via Hierarchical Topic Trajectory Model considering cross-modal correlations.
Proceedings of the IEEE International Conference on Acoustics, 2011

Multichannel harmonic and percussive component separation by joint modeling of spatial and spectral continuity.
Proceedings of the IEEE International Conference on Acoustics, 2011

Templatic features for modeling phoneme acquisition.
Proceedings of the 33th Annual Meeting of the Cognitive Science Society, 2011

Musical Instrument Identification Based on New Boosting Algorithm with Probabilistic Decisions.
Proceedings of the Speech, Sound and Music Processing: Embracing Research in India, 2011

2010
Harmonic and Percussive Sound Separation and Its Application to MIR-Related Tasks.
Proceedings of the Advances in Music Information Retrieval, 2010

Speech Spectrum Modeling for Joint Estimation of Spectral Envelope and Fundamental Frequency.
IEEE Trans. Speech Audio Process., 2010

SEMANTIC INDEXING AND KNOWN ITEM SEARCH BASED ON A UNIFIED MODEL WITH TOPIC TRANSITION REPRESENTATION.
Proceedings of the TRECVID 2010 workshop participants notebook papers, 2010

Analysis on speech characteristics for robust voice activity detection.
Proceedings of the 2010 IEEE Spoken Language Technology Workshop, 2010

Flexible Harmonic Temporal Structure for Modeling Musical Instrument.
Proceedings of the Entertainment Computing - ICEC 2010, 9th International Conference, 2010

A Roadmap Towards Versatile MIR.
Proceedings of the 11th International Society for Music Information Retrieval Conference, 2010

Autoregressive MFCC Models for Genre Classification Improved by Harmonic-percussion Separation.
Proceedings of the 11th International Society for Music Information Retrieval Conference, 2010

Multiple Pitch Transcription using DBN-based Musicological Models.
Proceedings of the 11th International Society for Music Information Retrieval Conference, 2010

Monophonic Instrument Sound Segregation by Clustering NMF Components Based on Basis Similarity and Gain Disjointness.
Proceedings of the 11th International Society for Music Information Retrieval Conference, 2010

Musical instrument identification based on harmonic temporal timbre features.
Proceedings of the ISCA Workshop on Statistical And Perceptual Audition, 2010

HMM-based approach for automatic chord detection using refined acoustic features.
Proceedings of the IEEE International Conference on Acoustics, 2010

Music mood classification by rhythm and bass-line unit pattern analysis.
Proceedings of the IEEE International Conference on Acoustics, 2010

Melody line estimation in homophonic music audio signals based on temporal-variability of melodic source.
Proceedings of the IEEE International Conference on Acoustics, 2010

R-means localization: A simple iterative algorithm for range-difference-based source localization.
Proceedings of the IEEE International Conference on Acoustics, 2010

A sparse component model of source signals and its application to blind source separation.
Proceedings of the IEEE International Conference on Acoustics, 2010

Designing the Wiener post-filter for diffuse noise suppression using imaginary parts of inter-channel cross-spectra.
Proceedings of the IEEE International Conference on Acoustics, 2010

Consistent Wiener Filtering: Generalized Time-Frequency Masking Respecting Spectrogram Consistency.
Proceedings of the Latent Variable Analysis and Signal Separation, 2010

Nonnegative Matrix Factorization with Markov-Chained Bases for Modeling Time-Varying Patterns in Music Spectrograms.
Proceedings of the Latent Variable Analysis and Signal Separation, 2010

Crystal-MUSIC: Accurate Localization of Multiple Sources in Diffuse Noise Environments Using Crystal-Shaped Microphone Arrays.
Proceedings of the Latent Variable Analysis and Signal Separation, 2010

Blind Estimation of Locations and Time Offsets for Distributed Recording Devices.
Proceedings of the Latent Variable Analysis and Signal Separation, 2010

2009
Note detection with dynamic bayesian networks as a postanalysis step for NMF-based multiple pitch estimation techniques.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2009

Blind alignment of asynchronously recorded signals for distributed microphone array.
Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2009

Orpheus: Automatic Composition System Considering Prosody of Japanese Lyrics.
Proceedings of the Entertainment Computing, 2009

Musical Bass-Line Pattern Clustering and Its Application to Audio Genre Classification.
Proceedings of the 10th International Society for Music Information Retrieval Conference, 2009

Minimum Classification Error Training to Improve Isolated Chord Recognition.
Proceedings of the 10th International Society for Music Information Retrieval Conference, 2009

Stereo-input speech recognition using sparseness-based time-frequency masking in a reverberant environment.
Proceedings of the INTERSPEECH 2009, 2009

Audio genre classification using percussive pattern clustering combined with timbral features.
Proceedings of the 2009 IEEE International Conference on Multimedia and Expo, 2009

Rhythm map: Extraction of unit rhythmic patterns and analysis of rhythmic structure from music acoustic signals.
Proceedings of the IEEE International Conference on Acoustics, 2009

Complex NMF: A new sparse representation for acoustic signals.
Proceedings of the IEEE International Conference on Acoustics, 2009

Extending Nonnegative Matrix Factorization - A discussion in the context of multiple frequency estimation of musical signals.
Proceedings of the 17th European Signal Processing Conference, 2009

2008
Specmurt Analysis of Polyphonic Music Signals.
IEEE Trans. Speech Audio Process., 2008

Sound Source Localization with Front-Back Judgement by Two Microphones Asymmetrically Mounted on a Sphere.
J. Multim., 2008

A Real-time Equalizer of Harmonic and Percussive Components in Music Signals.
Proceedings of the ISMIR 2008, 2008

Explicit consistency constraints for STFT spectrograms and their application to phase reconstruction.
Proceedings of the ISCA Tutorial and Research Workshop on Statistical and Perceptual Audition, 2008

Computational auditory induction by missing-data non-negative matrix factorization.
Proceedings of the ISCA Tutorial and Research Workshop on Statistical and Perceptual Audition, 2008

On-line handwritten Kanji string recognition based on grammar description of character structures.
Proceedings of the 19th International Conference on Pattern Recognition (ICPR 2008), 2008

Modulation analysis of speech through orthogonal FIR filterbank optimization.
Proceedings of the IEEE International Conference on Acoustics, 2008

Harmonic-Temporal-Timbral Clustering (HTTC) for the analysis of multi-instrument polyphonic music signals.
Proceedings of the IEEE International Conference on Acoustics, 2008

Auxiliary function approach to parameter estimation of constrained sinusoidal model for monaural speech separation.
Proceedings of the IEEE International Conference on Acoustics, 2008

A blind noise decorrelation approach with crystal arrays on designing post-filters for diffuse noise suppression.
Proceedings of the IEEE International Conference on Acoustics, 2008

Separation of a monaural audio signal into harmonic/percussive components by complementary diffusion on spectrogram.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

2007
Single and Multiple F<sub>0</sub> Contour Estimation Through Parametric Spectrogram Modeling of Speech in Noisy Environments.
IEEE Trans. Speech Audio Process., 2007

A Multipitch Analyzer Based on Harmonic Temporal Structured Clustering.
IEEE Trans. Speech Audio Process., 2007

Sound Source Localization by Asymmetrically Arrayed 2ch Microphones on a Sphere.
Proceedings of the IEEE 9th Workshop on Multimedia Signal Processing, 2007

Multipitch Analysis with Harmonic Nonnegative Matrix Approximation.
Proceedings of the 8th International Conference on Music Information Retrieval, 2007

Automatic Decision of Piano Fingering Based on a Hidden Markov Models.
Proceedings of the IJCAI 2007, 2007

Online Handwritten Kanji Recognition Based on Inter-stroke Grammar.
Proceedings of the 9th International Conference on Document Analysis and Recognition (ICDAR 2007), 2007

Rhythm and Tempo Analysis Toward Automatic Music Transcription.
Proceedings of the IEEE International Conference on Acoustics, 2007

Harmonic-Temporal Clustering of Speech for Single and Multiple F0 Contour Estimation in Noisy Environments.
Proceedings of the IEEE International Conference on Acoustics, 2007

Probabilistic Approach to Automatic Music Transcription from Audio Signals.
Proceedings of the IEEE International Conference on Acoustics, 2007

2006
Speech analyzer using a joint estimation model of spectral envelope and fine structure.
Proceedings of the INTERSPEECH 2006, 2006

Model Adaptation for Long Convolutional Distortion by Maximum Likelihood Based State Filtering Approach.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Effect of Learning on Listening to Ultra-Fast Synthesized Speech.
Proceedings of the 28th International Conference of the IEEE Engineering in Medicine and Biology Society, 2006

2005
Specmurt Analysis of Multi-Pitch Music Signals with Adaptive Estimation of Common Harmonic Structure .
Proceedings of the ISMIR 2005, 2005

Harmonic-Temporal Clustering via Deterministic Annealing EM Algorithm for Audio Feature Extraction.
Proceedings of the ISMIR 2005, 2005

Model adaptation by state splitting of HMM for long reverberation.
Proceedings of the INTERSPEECH 2005, 2005

Audio stream segregation of multi-pitch music signal based on time-space clustering using Gaussian kernel 2-dimensional model.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

2004
Galatea: Open-Source Software for Developing Anthropomorphic Spoken Dialog Agents.
Proceedings of the Life-like characters - tools, affective functions, and applications., 2004

Rhythm and Tempo Recognition of Music Performance from a Probabilistic Approach.
Proceedings of the ISMIR 2004, 2004

Complex spectrum circle centroid for microphone-array-based noisy speech recognition.
Proceedings of the INTERSPEECH 2004, 2004

Specmurt anasylis: a piano-roll-visualization of polyphonic music signal by deconvolution of log-frequency spectrum.
Proceedings of the ISCA Tutorial and Research Workshop on Statistical and Perceptual Audio Processing, 2004

Model composition by lagrange polynomial approximation for robust speech recognition in noisy environment.
Proceedings of the INTERSPEECH 2004, 2004

Multi-pitch trajectory estimation of concurrent speech based on harmonic GMM and nonlinear kalman filtering.
Proceedings of the INTERSPEECH 2004, 2004

Separation of harmonic structures based on tied Gaussian mixture model and information criterion for concurrent sounds.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

2003
Automatic rhythm transcription from multiphonic MIDI signals.
Proceedings of the ISMIR 2003, 2003

On-line Overlaid-Handwriting Recognition Based on Substroke HMMs.
Proceedings of the 7th International Conference on Document Analysis and Recognition (ICDAR 2003), 2003

Generation of Hierarchical Dictionary for Stroke-order Free Kanji Handwriting Recognition Based on Substroke HMM.
Proceedings of the 7th International Conference on Document Analysis and Recognition (ICDAR 2003), 2003

2002
Pen Pressure Features for Writer-Independent On-Line Handwriting Recognition Based on Substroke HMM.
Proceedings of the 16th International Conference on Pattern Recognition, 2002

Context-dependent substroke model for HMM-based on-line handwriting recognition.
Proceedings of the Eighth International Workshop on Frontiers in Handwriting Recognition, 2002

Jacobian joint adaptation to noise, channel and vocal tract length.
Proceedings of the IEEE International Conference on Acoustics, 2002

Hidden Markov model for automatic transcription of MIDI signals.
Proceedings of the IEEE 5th Workshop on Multimedia Signal Processing, 2002

2001
Dynamic Time-Alignment Kernel in Support Vector Machine.
Proceedings of the Advances in Neural Information Processing Systems 14 [Neural Information Processing Systems: Natural and Synthetic, 2001

Support vector machine with dynamic time-alignment kernel for speech recognition.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

Substroke Approach to HMM-Based On-line Kanji Handwriting Recognition.
Proceedings of the 6th International Conference on Document Analysis and Recognition (ICDAR 2001), 2001

Multiple-regression hidden Markov model.
Proceedings of the IEEE International Conference on Acoustics, 2001

2000
Speaker adaptation of acoustic models using correlations of training transfer vectors.
Syst. Comput. Jpn., 2000

IPA Japanese Dictation Free Software Project.
Proceedings of the Second International Conference on Language Resources and Evaluation, 2000

Jacobian adaptation of HMM with initial model selection for noisy speech recognition.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Feature-dependent allophone clustering.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Free software toolkit for Japanese large vocabulary continuous speech recognition.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Asynchronous-transition HMM.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
An address data entry system with a multimodal interface including speech recognition.
Syst. Comput. Jpn., 1999

1998
Two-step generation of variable-word-length language model integrating local and global constraints.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

1997
ASR and TTS telecommunications applications in Japan.
Speech Commun., 1997

Speech recognition and synthesis technology development at NTT for telecommunications services.
Int. J. Speech Technol., 1997

Vector-field-smoothed Bayesian learning for fast and incremental speaker/telephone-channel adaptation.
Comput. Speech Lang., 1997

Fast adaptation of acoustic models to environmental noise using jacobian adaptation algorithm.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

Variable-length language modeling integrating global constraints.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

Discrete mixture HMM.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997

Jacobian approach to fast acoustic model adaptation.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997

Improved estimation of supervision in unsupervised speaker adaptation.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997

1996
A speaker-adaptation technique for context-dependent models represented by hidden markov networks.
Syst. Comput. Jpn., 1996

Speaker-independent speech recognition based on tree-structured speaker clustering.
Comput. Speech Lang., 1996

LR-parser-driven viterbi search with hypotheses merging mechanism using context-dependent phone models.
Proceedings of the 4th International Conference on Spoken Language Processing, 1996

Iterative unsupervised speaker adaptation for batch dictation.
Proceedings of the 4th International Conference on Spoken Language Processing, 1996

Minimum classification error training for a small amount of data enhanced by vector-field-smoothed Bayesian learning.
Proceedings of the 1996 IEEE International Conference on Acoustics, 1996

Tied-structure HMM based on parameter correlation for efficient model training.
Proceedings of the 1996 IEEE International Conference on Acoustics, 1996

1995
Interactive voice technology development for telecommunications applications.
Speech Commun., 1995

Speaker-Consistent Parsing for Speaker-Independent Continuous Speech Recognition.
IEICE Trans. Inf. Syst., 1995

Unsupervised Speaker Adaptation Using All-Phoneme Ergodic Hidden Markov Network.
IEICE Trans. Inf. Syst., 1995

Automatic Determination of the Number of Mixture Components for Continuous HMMs Based a Uniform Variance Criterion.
IEICE Trans. Inf. Syst., 1995

Speech Recognition Using Function-Word <i>N</i>-Grams and Content-Word <i>N</i>-Grams.
IEICE Trans. Inf. Syst., 1995

Fast and accurate beam search using forward heuristic functions in HMM-LR speech recognition.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

Syllabic duration control for vocabulary-free speech recognition.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

Vector-field-smoothed Bayesian learning for incremental speaker adaptation.
Proceedings of the 1995 International Conference on Acoustics, 1995

Four-level tied-structure for efficient representation of acoustic modeling.
Proceedings of the 1995 International Conference on Acoustics, 1995

On the use of scalar quantization for fast HMM computation.
Proceedings of the 1995 International Conference on Acoustics, 1995

1994
Telephone line characteristic adaptation using vector field smoothing technique.
Proceedings of the 3rd International Conference on Spoken Language Processing, 1994

Tree-structured speaker clustering for speaker-independent continuous speech recognition.
Proceedings of the 3rd International Conference on Spoken Language Processing, 1994

All-phoneme ergodic hidden Markov network for unsupervised speaker adaptation.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

Tree-structured speaker clustering for fast speaker adaptation.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1993
Suprasegmental duration control with matrix parsing in continuous speech recognition.
Speech Commun., 1993

Feature extraction using a matrix coefficient filter for speech recognition.
Speech Commun., 1993

A neural fuzzy training approach for improving speech recognition.
Syst. Comput. Jpn., 1993

ATREUS: a speech recognition front-end for a speech translation system.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

The possibility for acquisition of statistical network grammar using ergodic HMM.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

ATR's speech translation system: ASURA.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

A dynamic approach to speaker adaptation of hidden Markov networks for speech recognition.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

Speech recognition using particle n-grams and content-word n-grams.
Proceedings of the Third European Conference on Speech Communication and Technology, 1993

Spoken Language Translation System.
Proceedings of the 13th International Joint Conference on Artificial Intelligence. Chambéry, France, August 28, 1993

Matrix parser and its application to HMM-based speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 1993

ATREUS: a comparative study of continuous speech recognition systems at ATR.
Proceedings of the IEEE International Conference on Acoustics, 1993

Rapid speaker adaptation using speaker-mixture allophone models applied to speaker-independent speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 1993

1992
Continuous mixture HMM-LR using the a* algorithm for continuous speech recognition.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Appropriate error criterion selection for continuous speech HMM minimum error training.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Speaker adaptation based on transfer vector field smoothing with continuous mixture density HMMs.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

The SSS-LR continuous speech recognition system: integrating SSS-derived allophone models and a phoneme-context-dependent LR parser.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Hardware implementation of realtime 1000-word HMM-LR continuous speech recognition.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Enhancement of ATR's spoken language translation system: SL-TRANS2.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Continuously spoken sentence recognition by HMM-LR.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Vector field smoothing principle for speaker adaptation.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

A successive state splitting algorithm for efficient allophone modeling.
Proceedings of the 1992 IEEE International Conference on Acoustics, 1992

Pitch dependent phone modelling for HMM based speech recognition.
Proceedings of the 1992 IEEE International Conference on Acoustics, 1992

1991
A matrix representation of HMM-based speech recognition algorithms.
Proceedings of the Second European Conference on Speech Communication and Technology, 1991

Phoneme-context-dependent LR parsing algorithms for HMM-based continuous speech recognition.
Proceedings of the Second European Conference on Speech Communication and Technology, 1991

A pairwise discriminant approach to robust phoneme recognition by time-delay neural networks.
Proceedings of the 1991 International Conference on Acoustics, 1991

Phoneme recognition by phoneme filter neural networks.
Proceedings of the 1991 International Conference on Acoustics, 1991

1990
Phoneme recognition by pairwise discriminant TDNNs.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Isolated word recognition using pitch pattern information.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Estimation of unknown context using a phoneme environment clustering algorithm.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Sentence speech recognition using semantic dependency analysis.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Speaker weighted training of HMM using multiple reference speakers.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Line spectrum pair frequency - based distance measures for speech recognition.
Proceedings of the First International Conference on Spoken Language Processing, 1990

Statistical study on voice individuality conversion across different languages.
Proceedings of the First International Conference on Spoken Language Processing, 1990

A continuous speech recognition system based on a two-level grammar approach.
Proceedings of the 1990 International Conference on Acoustics, 1990

1989
Phoneme environment clustering for speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 1989

1986
Duality theory of composite sinusoidal modeling and linear prediction.
Proceedings of the IEEE International Conference on Acoustics, 1986


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