SHIKANO Kiyohiro | Graduate School of Information Science, Nara Institute of Science and Technology

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概要

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著作論文

Building an Effective Speech Corpus by Utilizing Statistical Multidimensional Scaling Method
Cost Reduction of Acoustic Modeling for Real-Environment Applications Using Unsupervised and Selective Training
Cross-language Voice Conversion Evaluation Using Bilingual Databases (特集音声言語情報処理とその応用)
Effect of Central Limit Theorem non-compliance on blind separation of speech by negentropy maximization
Robots that can hear, understand and talk
Probability Distribution of Time-Series of Speech Spectral Components(Audio/Speech Coding)(Applications and Implementations of Digital Signal Processing)
A design of adaptive beamformer based on average speech spectrum for noisy speech recognition
A Microphone Array-Based 3-D N-Best Search Method for Recognizing Multiple Sound Sources
3D N-best 探索法に基づく複数音源の位置推定と音声認識の統合
複数話者の音声認識における音源方向経路間距離を用いた3-D N-best探索法の評価
Non-Audible Murmur (NAM) Recognition(2004 IEICE Excellent Paper Award)
Non-Audible Murmur (NAM) Recognition Exploiting Adaptation Techniques
Objective sound quality evaluation for combination method of beamforming and spectral subtraction (応用音響)
Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming
Rapid Compensation of Temperature Fluctuation Effect for Multichannel Sound Field Reproduction System
Development, Long-Term Operation and Portability of a Real-Environment Speech-Oriented Guidance System
On-Line Relaxation Algorithm Applicable to Acoustic Fluctuation for Inverse Filter in Multichannel Sound Reproduction System(Sound Field Reproduction, Multi-channel Acoustic Signal Processing)
Iterative Inverse Filter Relaxation Algorithm for Adaptation to Acoustic Fluctuation in Sound Reproduction System
Sound Reproduction System Including Adaptive Compensation of Temperature Fluctuation Effect for Broad-Band Sound Control(Special Section on Digital Signal Processing)
Elderly Acoustic Models for Large Vocabulary Continuous Speech Recognition
Interface for Barge-in Free Spoken Dialogue System Combining Adaptive Sound Field Control and Microphone Array(Speech and Hearing)
A Self-Generator Method for Initial Filters of SIMO-ICA Applied to Blind Separation of Binaural Sound Mixtures(Blind Source Separation, Multi-channel Acoustic Signal Processing)
Multistage SIMO-Model-Based Blind Source Separation Combining Frequency-Domain ICA and Time-Domain ICA(Adaptive Signal Processing and Its Applications)
Direction of Arrival Estimation Using Nonlinear Microphone Array
Evaluation of Extremely Small Sound Source Signals Used in Speaking-Aid System with Statistical Voice Conversion

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