SARUWATARI Hiroshi | Nara Institute of Science and Technology

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概要

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著作論文

Development of real-time audio localization control system (応用音響)
EA2010-24 Development of real-time audio localization control system
Sound reproduction based on multi-channel inverse filtering and WFS
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
Reducing Computation Time of the Rapid Unsupervised Speaker Adaptation Based on HMM-Sufficient Statistics(Speech and Hearing)
Improving Rapid Unsupervised Speaker Adaptation Based on HMM-Sufficient Statistics in Noisy Environments Using Multi-Template Models(Speech Recognition, Statistical Modeling for Speech Processing)
Utterance-Based Selective Training for the Automatic Creation of Task-Dependent Acoustic Models(Speech Recognition, Statistical Modeling for Speech Processing)
Designing Target Cost Function Based on Prosody of Speech Database(Speech Synthesis and Prosody, Corpus-Based Speech Technologies)
Designing Target Cost Function Based on Prosody of Speech Database
A MAP Estimator for the Enhancement of Speech Signal Separated by ICA Algorithm (国際ワークショップ Frontiers in Speech and Hearing Research)
Effect of Central Limit Theorem non-compliance on blind separation of speech by negentropy maximization
Blind Separation of Speech by Fixed-Point ICA with Source Adaptive Negentropy Approximation(Blind Source Separation, Multi-channel Acoustic Signal Processing)
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)
Non-Audible Murmur (NAM) Recognition Exploiting Adaptation Techniques
Development and evaluation of pocket-size real-time blind source separation microphone
Objective sound quality comparison based on higher-order statistics for nonlinear noise reduction methods (応用音響)
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
Interface for Barge-in Free Spoken Dialogue System Using Nullspace Based Sound Field Control and Beam forming (Speech/Audio Processing, Multidimensional Signal Processing and Its Application)
On-Line Relaxation Algorithm Applicable to Acoustic Fluctuation for Inverse Filter in Multichannel Sound Reproduction System(Sound Field Reproduction, Multi-channel Acoustic Signal Processing)
複数モデルを用いた十分統計量に基く教師なし話者適応における学習話者のクラス化の検討
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)
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)
Evaluation of Extremely Small Sound Source Signals Used in Speaking-Aid System with Statistical Voice Conversion
Improvements of the One-to-Many Eigenvoice Conversion System
Esophageal Speech Enhancement Based on Statistical Voice Conversion with Gaussian Mixture Models
Adaptive Training for Voice Conversion Based on Eigenvoices
Blind Separation and Deconvolution for Convolutive Mixture of Speech Combining SIMO-Model-Based ICA and Multichannel Inverse Filtering(Engineering Acoustics)
High-Fidelity Blind Separation of Acoustic Signals Using SIMO-Model-Based Independent Component Analysis(Engineering Acoustics)
Overdetermined Blind Separation for Real Convolutive Mixtures of Speech Based on Multistage ICA Using Subarray Processing(Speech/Acoustic Signal Processing)(Digital Signal Processing)
Stable Learning Algorithm for Blind Separation of Temporally Correlated Acoustic Signals Combining Multistage ICA and Linear Prediction(Digital Signal Processing)
Blind Source Separation of Acoustic Signals Based on Multistage ICA Combining Frequency-Domain ICA and Time-Domain ICA
Fast-Convergence Algorithm for Blind Source Separation Based on Array Signal Processing
An Iterative Inverse Filter Design Method for the Multichannel Sound Field Sound Field Reproduction System(Special Section on Acoustic Signal Processing)
Sound Field Reproduction by Wavefront Synthesis Using Directly Aligned Multi Point Control
Theoretical Analysis of Amounts of Musical Noise and Speech Distortion in Structure-Generalized Parametric Blind Spatial Subtraction Array
Speech Prior Estimation for Generalized Minimum Mean-Square Error Short-Time Spectral Amplitude Estimator
Comparison of Methods for Topic Classification of Spoken Inquiries
Robust Sound Field Reproduction against Listeners Movement Utilizing Image Sensor

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