M.S. and Ph.D. from Japan Advanced Institute of Science and Technology (1996,1999) 北陸先端科学技術大学院大学
2000 - 2001 : 日本学術振興会特別研究員（DC2）（1998），ATR人間情報通信研究所 客員研究員（1999），ケンブリッジ大学CNBH客員研究員（2000-2001），日本学術振興会特別研究員（PD，北陸先端科学技術大学院大学 情報科学研究科）（1999-2001）
2000 - 2001 : JSPS Research Fellow (DC2)(1998), Visiting researcher, ATR Human Information Processing Laboratories (1999), Visiting Associate, CNBH, Univ. of Cambridge (2000-2001), JSPS Research Fellow (PD)(1999-2001)
Intelligent robotics, Perceptual information processing, Intelligent informatics
Audio Information Hiding, 音声信号処理, 聴覚情景解析, 聴覚モデル, Speech dereverberation, Computational Auditory Scene Analysis, Auditory filterbank
Construction of the auditory filterbank
The aim of this work is to construct the auditory filterbank that can account psycho-acoustical data and physiological data for frequency selectivity of the auditory system.
Computational Auditory Scene Analysis
The study of computational theory of the auditory system tries to answer the following questions:
Extraction of the fundamental frequency of speech in real environments
Extraction of the fundamental frequency (F0) of target speech is an important problem not only in speech analysis/synthesis but also in various speech signal processings such as speech segregation. Various F0 estimation methods have been proposed, but the most of these methods have the drawbacks for estimating accurate F0s of target speech in real environments. My approach is to construct an estimation model based on computational auditory scene analysis (CASA).
A study on the speech dereverberation method
To dereverberate the original signal from a reverberant signal is an important issue concerning speech signal processing such as preprocessing for speech recognition systems. Most of the inverse filtering methods have to measure the impulse response of the room acoustics to determine its inverse filter before the dereverberation. Moreover, the impulse response temporally varies with various environmental factors (temperature etc.), so the room acoustics have to be measured each time these methods are used. In this work, it has been trying to model a speech dereverberation based on the Modulation Transfer Function, without measuring the impulse response of room acoustics.
Discrete Signal Processing, Human Perceptual Systems and its Models, Statistical Signal Processing, 離散信号処理特論, 知覚情報処理特論, 統計的信号処理特論
信号処理学会, International Speech Communication Association, 電子情報通信学会, 日本音響学会, Research Institute of Signal Processing, Japan, Information and Communication Engineers, Institute of Electronics, The Acoustical Society of Japan, Institute of Electrical and Electronics Engineers, Acoustical Society of America
2017 International Workshop on Nonlinear Circuits and Signal Processing (NCSP17) Committee member (General Chair) , JAIST, Prof. Unoki Masashi , 2017 - 2017 , Guam, USA
2016 International Workshop on Nonlinear Circuits and Signal Processing (NCSP16) Committee member (General Vice Chair) , RISP , 2016 - 2016 , Honolulu, Hawaii, USA
The 14th IWDW, International Workshop on Digital-forensics and Watermarking (IWDW 2015) , Organizaing committee, , 2015 - 2015 , Tokyo University of Sciences
・ Best paper award , Masashi Unoki , 11th International Conference on Social Computing and Social Media (SCSM 2019) , 2019
・ Best paper award , Masashi Unoki , 14th International Conference on Intelligent Information Hiding and Multimedia Signal Processing (II , 2018
・ 平成30年度支部学会活動貢献賞 , 日本音響学会北陸支部 , 2018