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SHIMAMURA Tetsuya

■Faculty: Graduate School of Science and Engineering ■TEL:
■Position: Professor ■FAX:
■Address: 255 Shimo-Okubo, Sakura-ku, Saitama City, Saitama 338-8570, JAPAN ■Mail Address:
■Web site:

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Faculty of Engineering

Academic Societies

Academic Societies
IEEE
EURASIP

Research

Books, Articles, etc.

Books
MATLABによる実戦ディジタル信号処理
トリケップス:121 2010
島村徹也

知識ベｰスβ版 1群5編(信号理論)4章 定常スペクトル解析
電子情報通信学会:11 2010
島村徹也を含む多数

知識ベｰスβ版 1群5編(信号理論)5章 一次元ウイナｰフィルタ
電子情報通信学会:5 2010
島村徹也を含む多数

物理学辞典　（三訂版）
培風館:1 200509
島村徹也

MATLABマルチメディア信号処理　下　音声・画像・通信
培風館:1-110 2004
池原雅章,島村徹也,真田幸俊

MATLABマルチメディア信号処理　上　ディジタル信号処理の基礎
培風館 2004
池原雅章,島村徹也

MATLAB プログラム事例解説　Ⅰ　音声通信－特徴抽出と雑音低減－
トリケップス 2001
島村徹也

一次元ディジタル信号処理の基礎
培風館 2001
高橋進一，島村徹也

Articles
A Joint Iterative Estimation of Noise Variance and AR Parameters
,International Journal of Information Sciences and Computer Engineering,2(1):1-6 2011
J. Gamba, T. Shimamura, S. Kawasaki, M. Higuchi and H. Murakami

Extended Fundamental Frequency Extraction Using Exponentiated Amplitude Spectrum with Band-Limitation
,Proceedings of IEEE International Conference on Signal Acquisition and Processing:V1-168-V1-172 2011
S. Motegi and T. Shimamura

Iterative Edge-Preserving Adaptive Wiener Filter for Image Denoising
,Proceedings of IEEE International Conference on Signal Acquisition and Processing:V1-168-V1-172 2011
C. Abe and T. Shimamura

Noise Estimation Using Only Current Frame of Speech for Spectral Subtraction
,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:425-428 2011
A. Matsukawa and T. Shimamura

Noise Reduction Based on Pitch Synchronous Addition and Subtraction for LPC Analysis
,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:348-351 2011
L. Liu and T. Shimamura

Performance Improvement of MUSIC by AR Model Based Data Extension
,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:276-279 2011
T. Yokose and T. Shimamura

Transform Domain Sato Algorithm for Blind Channel Equalization
,Journal of Signal Processing,15(2):123-131 2011
M. L. R. Khan and T. Shimamura

A Lattice Based Fast Adaptive Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:624-627 2010
K.Tashiro and T.Shimamura

A New Variable Step Size for Normalized LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:584-587 2010
T.Arnantapunpong, T Shimamura and S.A.Jimaa

A Thresholding-Based Image Denoising Method
,Proceedings of World Academy of Science and Technology,65:911-916 2010
S. Suhaila and T. Shimamura

Accuracy Improvement of Speaker Authentication in Noisy Environments Using Bone-Conducted Speech
,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:197-200 2010
N. Yamasaki and T. Shimamura

Autocorrelation and Double Autocorrelation Based Spectral Representations for Word Recognition in Noisy Environments
,信号処理シンポジウム講演論文集,B-10-2:475-478 2010
T. Shimamura and N. D. Nguyen

Autocorrelation and Double Autocorrelation Representations for a Noisy Word Recognition System
,Proceedings of INTERSPEECH 2010:1712-1715 2010
T. Shimamura and N. N. Dinh

Convergence Evaluation of a Random Step-Size NLMS Adaptive Algorithm in System Identification
,Proceedings of IEEE International Conference on Signal Processing:135-138 2010
S. A. Jimaa and T. Shimamura

Digital Notch Filter for Noise Reduction in Automatic Train Control System
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:536-539 2010
H.Takekawa, T.Shimamura and S.Irie

Noise Estimation Based on Series Expansion of Orthogonal Functions
,Proceedings of IEEE International Conference on Signal Processing:115-118 2010
T. Akasaka and T. Shimamura

Noise Spectrum Estimation Based on Optimum Smoothing for Robust Speech Enhancement
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:293-296 2010
A.Saha, T.Shimamura

Pitch Characteristics of Bone Conducted Speech
,Proceeding of European Signal Processing Conference (EUSIPCO):795-798 2010
M. S. Rahman and T. Shimamura

Pitch Determination Using Autocorrelation Function in Spectral Domain
,Proceedings of INTERSPEECH 2010:653-656 2010
M. S. Rahman and T. Shimamura

Pitch Determination Using Windowless Autocorrelation Based Cepstrum Method
,Proceedings of APSIPA Annual Summit and Conference 2010:514-517 2010
M. A. F. M. Rashidul Hasan, M. Shahidur Rahman and T. Shimamura

Power Spectrum Estimation Method for Image Denoising by Frequency Domain Wiener Filter
IEEE,Proceedings of IEEE International Conference on Computer and Automation Engineering:608-612 2010
S.Suhaila and T.Shimamura

Time-Varying Channel Estimation Using Amplitude-Division Based Parallel NLMS Technique
,Proceedings of IEEE International Conference on Wireless and Mobile Computing, Networking and Communications:580-585 2010
R. Yasmin and T. Shimamura

Tracking by Nonuniform Amplitude Division Based LMS Algorithm for Time Varying Channels
,Proceedings of IEEE International Symposium on Circuits and Systems:2852-2855 2010
R. Yasmin and T. Shimamura

Utilization of Adaptive Line Enhancement and Noise Compensation for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:524-527 2010
C.Tiengwattanatum and T.Shimamura

研究室紹介島村研究室(埼玉大学)
,Journal of Signal Processing,14(3):201-209 2010
島村徹也

双相関､差分､べき乗を利用した雑音混入音声の基本周波数抽出ルゴリズム
,電子情報通信学会技術研究報告,SP2010-113:65-69 2010
成田雅俊, 島村徹也

Equalization of Time-Variant Communications Channels via Adaptive AR Prefiltering
WRI,Proceedings of WRI World Congress on Computer Science and Information Engineering:327-331 200903
Shimamura T.

Dual Adaptive Pre-Whitening Filters for LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:1-4 200903
Tashiro K. and Shimamura T.

Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:165-168 200903
Nakamura N., Ikeda T. and Shimamura T.

Image Restoration via Wiener Filtering with Improved Noise Estimation
WSEAS,Proceedings of WSEAS International Conference on Signal Processing, Robotics and Automation:315-320 200902
Furuya H., Eda S. and Shimamura T.

Blind Channel Equalization with Amplitude Banded Godard and Sato Algorithms
Academy Publisher,Journal of Communications,4(6):388-395 2009
M.L.R.Kahn, M.H.Wondimagegnehu and T.Shimamura

Convergence Evaluation of Variable Step-Size NLMS Algorithms in Adaptive Channel Equalization
IEEE,Proceedings of IEEE International Symposium on Signal Processing and Information Technology:145-150 2009
S.A.Jimaa, A.Al-Simiri, R.M.Shubair and T.Shimamura

Image Restoration via Wiener Filtering in the Frequency Domain
,WSEAS Transactions on Signal Processing,5(Issue 2):63-73 2009
Furuya H., Eda S. and Shimamura T.

Reverberated Speech Enhancement Using Neural Networks
IEEE,Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems:441-444 2009
B.D.Dufera and T.Shimamura

ラティスフィルタを用いた高速適応アルゴリズム
電子情報通信学会,電子情報通信学会技術研究報告,109(226):17-22 2009
田代和義、島村徹也

音声の2重自己相関関数のスペクトル表現と雑音環境下単語認識システムへの応用
電子情報通信学会,第11回音声言語シンポジウム講演論文集:135-140 2009
グエン・ゴック・ディン、島村徹也

画像復元のための反復的エッジ保存適応ウィナーフィルタ
電子情報通信学会,電子情報通信学会技術研究報告,109(226):23-28 2009
安部ちかこ、島村徹也

振幅スペクトルのべき乗引き算に基づく音声の基本周波数抽出
電子情報通信学会,電子情報通信学会技術研究報告,109(99):93-98 2009
茂木沙織、島村徹也

直交関数系の級数展開に基づくスペクトル引き算
電子情報通信学会,第11回音声言語シンポジウム講演論文集:129-134 2009
赤坂泰司、島村徹也

反復相互相関関数を用いた時間遅延推定
電子情報通信学会,電子情報通信学会論文誌,J92-A(9):651-655 2009
中村尚之、島村徹也

Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization
IEEE,Proceedings of International Conference on Computer and Information Technology:768-772 200812
Khan, M.L.R., Wondimagegnehu M.H. and Shimamura T.

High Pitch Source Isolation Using Complex Cepstrum in the Autocorrelation Domain
IEEE,Proceedings of IEEE Asia Pacific Conference on Circuits and Systems:1284-1287 200812
Derebssa B. and Shimamura T.

An Efficient and Effective Variable Step Size NLMS Algorithm
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:1640-1643 200810
Takekawa H., Shimamura T. and Jimaa S.

Adaptive Line Enhancer for Time Delay Estimation of Ultrasonic Echoes
IEEE,Proceedings of International Symposium on Communication and Information Technology:368-371 200810
Tiengwattanatum C., Nakamura N. and T., Shimamura T.

Amplitude-Division Parallel LMS Estimator
IEEE,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:950-953 200808
Shimamura T., Oikawa S. and Tsuda Y.

Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm
IEEE,Proceedings of IEEE International Networking and Communications Conference:23-26 200805
Jimaa S., Shimamura T. and Takekawa H.

Iterative Cross-Correlation Method for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:387-390 200803
Nakamura N. and Shimamura T.

Wavelet Based Denoising for Images Degraded by Poisson Noise
IASTED,Proceedings of IASTED International Conference on Biomedical Engineering:436-440 200802
Shimamura T., Eda S., Ito T., Kuwano Y. and Takahashi Y.

Special Section on Nonlinear Circuits and Signal Processing Editor's Note
,Journal of Signal Processing,12(6):412-413 2008
Shimamura T.

Special Section on Papers Awarded the Student Paper Award at NCSP'08 Editor's Note
,Journal of Signal Processing,12(4):269-270 2008
Shimamura T.

線形予測誤差を用いた骨導音声の品質改善
電子情報通信学会,信号処理シンポジウム講演論文集:164-169 2008
杉山貴紀,島村徹也,八嶋弘幸

双対の適応白色化フィルタを用いたLMSアルゴリズム
電子情報通信学会,信号処理シンポジウム講演論文集:195-200 2008
田代和義,島村徹也

騒音環境下に有効な骨導・気導一体型マイクロホン
,ナノ材料・IT新技術説明会資料集:11-16 2008
島村徹也

Amplitude Banded Sato Algorithm for Blind Channel Equalization
IEEE,Proceedings of IEEE International Conference on Signal Processing and Communications:1463-1466 200711
Khan M.L.R., Mohammed H.W. and Shimamura T.

Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
NOLTA,Proceedings of International Symposium on Nonlinear Theory and Its Applications:441-444 200709
Eda S. and Shimamura T.

Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
IASTED,Proceedings of IASTED International Conference on Signal and Image Processing:576-205 200708
Shimamura T.

Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
IEEE,Proceedings of IEEE International Workshop on Nonlinear Dynamics of Electronic Systems:289-292 200707
Shimamura T.

Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
WSEAS,Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications:125-129 200701
Tanaka H., Ohhashi Y. and Shimamura T.

Discrete Cosine Transform Domain Parallel LMS Equalizer
WSEAS,Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications:119-124 200701
Mohammed H.W. and Shimamura T.

A Parallel Equalizer with LMS Adaptation in Discrete Cosine Transform Domain
,WSEAS Transactions on Signal Processing,2 2007
Mohammed H.W. and Shimamura T.

Bone-Conducted Speech for Speaker Verification
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:172-175 2007
Iijima S. and Shimamura T.

Complementary and Phase Banded LMS Equalizers for Rapidly Time-Varying Channels
,Journal of Signal Processing,11:51-60 2007
Mohammed H.W. and Shimamura T.

Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
,International Journal of Asia Digital Art and Design,6:11-18 2007
Xin W., Kondo K., Tateno K., Konma T. and Shimamura T.

Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
NICOGRAPH,Proceedings of NICOGRAPH 2007
Xin W., Kondo K., Tateno K., Konma T. and Shimamura T.

Indirect Cross-Correlation Method for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:74-77 2007
Nakamura N. and Shimamura T.

Linear Prediction Using Refined Autocorrelation Function
,EURASIP Journal on Audio, Speech and Music Processing,2007:Article ID 45962, 9 Pages 2007
Rahman M.S. and Shimamura T.

Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:213-216 2007
Tiengwattanatum C., Mohammed H.W. and Shimamura T.

Performance of Adaptive Nonlinear Predictor with Order Statistics in Impulsive Noise
,WSEAS Transactions on Signal Processing,2 2007
Tanaka H., Ohhashi Y. and Shimamura T.

Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:291-294 2007
Sugiyama T., Shimamura T. and Yashima H.

Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:583-586 2007
Kamata K., Yamashita K., Shimamura T. and Furukawa T.

音声信号の非線形予測と信号表現に関する研究
埼玉大学総合研究機構,総合研究機構研究プロジェクト研究成果報告書,5 (18年度):602-603 2007
島村徹也
Study on Nonlinear Prediction of Speech Signals and Signal Representation
リポジトリ

可変ステップサイズ正規化LMSアルゴリズムの一提案
,信号処理シンポジウム講演論文集:466-471 2007
竹川英樹,島村徹也

効率的な時間遅延推定のための間接的差分関数法
,信号処理シンポジウム講演論文集:401-405 2007
中村尚之,島村徹也

癒し音楽における1/fゆらぎと高周波成分との関連性
,音楽音響研究会資料,26:25-30 2007
島村徹也,小花あゆみ

A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels
IEEE,Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems:845-848 200612
Oikawa S.，Tsuda Y. and Shimamura T.

Adaptive Non-Linear Prediction with Order Statistics for Speech Signals in Mixture Noise
IEEE,Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems:295-298 200612
Tanaka H.，Ohhashi Y. and Shimamura T.

Active Noise Control Using A Refined Filtering Approach
,Proceedings of the 35th International Congress and Exposition on Noise Control Engineering 200612
Isozaki K.，Tsuda Y. and Shimamura T.

Wavelet Based Keyframe Extraction Method from Motion Capture Data
ADADA,Proceedings on Asia Digital Art and Design Association:128-129 200612
Xin W., Kondo K., Tateno K., Konma T. and Shimamura T.

Learning for Bone-Conducted Speech via Adaptive and Neural Filters
IEEE,Proceedings of International Conference on Signals and Electronic Systems 200609
Shimamura T. and Tamiya T.

A Harsh Noise Assessment Measure for Speech Enhancement
EURASIP,Proceedings of European Conference on Signal Processing 200609
Yamashita K. and Shimamura T.

Improving Bone-Conducted Speech Quality via Neural Network
IEEE,Proceedings of IEEE International Symposium on Signal Processing and Information Technology:628-632 200608
Shimamura T.，Mamiya J. and Tamiya T.

Speech Analysis Based on Modeling the Effective Voice Source(Speech Analysis, <Special Section> Statistical Modeling for Speech Processing)
電子情報通信学会,IEICE transactions on information and systems,E89-D(3):1107-1115 200603
島村徹也
リポジトリ

A Study on Normalized LMS Algorithm Using Refined Filtering Technique
WSEAS,Proceedings of 5th WSEAS International Conference on Signal Processing, Robotics and Automation:264-267 200602
Tsuda Y. and Shimamura T.

Coefficients--Delay Simultaneous Adaptation Scheme for Linear Equalization of Nonminimum Phase Channels(Digital Signal Processing)
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E89-A(1):248-259 200601
島村徹也
リポジトリ

Noise Estimation Using Multifrequency Regions for Spectral Subtraction
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Yamashita K. and Shimamura T.

Noise Estimation Using Multifrequency Regions for Spectral Subtraction
,Journal of Signal Processing,10:275-278 2006
Yamashita T. and Shimamura T.

Pitch Determination using Aligned AMDF
ISCA,Proceedings of International Conference on Spoken Language Processing:1714-1717 2006
Rahman M.S.，Tanaka H. and Shimamura T.

Refined Filtering for Normalized LMS Algorithm
,WSEAS Transactions on Signal Processing,2:261-264 2006
Tsuda Y. and Shimamura T.

高性能アクティブノイズキャンセルマイクロフォン開発におけるリアルタイムなディジタル雑音除去の研究
埼玉大学総合研究機構地域共同研究センター産学連携推進部門,埼玉大学地域共同研究センター紀要,7:64-64 2006
島村徹也,和田存功
A spectral subtraction technique is carried out in which noise is estimated for non-speech duration and the estimated noise spectrum is subtracted from the noisy speech spectrum for speech duration.
Noise estimation, Spectral subtraction
リポジトリ

A Refined Filtering Approach to Adaptive Line Enhancement
,回路とシステム軽井沢ワークショップ講演論文集:141-146 2006
Tsuda Y. and Shimamura T.

Active Noise Control Using Cascaded Adaptive Filters
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Isozaki K., Tsuda Y. and Shimamura T.

Active Noise Control Using Cascaded Adaptive Filters
,Journal of Signal Processing,10:279-282 2006
Isozaki K.，Tsuda Y. and Shimamura T.

Adaptive Time Variant Channel Equalization Using Phase Banded LMS Algorithm
,Journal of Signal Processing,10:227-230 2006
Mohammed H.W. and Shimamura T.

Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Mohammed H. W. and Shimamura T.

Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
,Proceedings of International Conference on Signals and Electronic Systems 2006
Rahman M.S.，Tanaka H. and Shimamura T.

Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
WSEAS,Proceedings of WSEAS International Conference on Electronics, Control and Signal Processing:47-52 200511
Shimamura T.，Yamauchi J.

反復処理を利用した改良スペクトル引き算(音声, <小特集>スマート信号処理とその画像・音声処理への応用論文)
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J88-A(11):1246-1257 200511
山下浩平,緒方伸哉,島村徹也
リポジトリ

Speech Enhancement Using a Technique of Adaptive Bias Suppression
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:540-544 200510
Tanaka H. and Shimamura T.

Voice Source Modeling for Accurate Speech Analysis
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:305-309 200510
Rahman M.S. and Shimamura T.

Quality Improvement of Bone-Conducted Speech
IEEE,Proceedings of European Conference on Circuit Theory and Design 200508
Shimamura T. and Tomikura T.

A Reconstruction Filter for Bone-Conducted Speech
IEEE,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:1847-1850 200508
Shimamura T. and Tamiya T.

Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:6288-6291 200505
Kobayashi T.，Shimamura T.，Hosoya T. and Takahashi Y.

Linear Prediction Using Homomorphic Deconvolution in the Autocorrelation Domain
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:2855-2858 200505
Rahman M.S. and Shimamura T.

Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
IEEE, EURASIP,Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing:6-9 200505
Gamba J. and Shimamura T.

Equalizer-Aided Time Delay Tracking Based on L_1-Normed Finite Differences(Digital Signal Processing)
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E88-A(4):978-987 200504
Gamba J. and Shimamura T.
リポジトリ

Spectrum Estimation by Noise-Compensated Data Extrapolation(Digital Signal Processing)
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E88-A(3):702-711 200503
Gamba J., Shimamura T.
リポジトリ

A Parallel Estimator with LMS Adaptation for Fast Fading Channels
,Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology:2C-12.4 2005
Oikawa S.，Tsuda Y. and Shimamura T.

Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
IEEE, EURASIP,Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing:404-409 2005
Tanaka H. and Shimamura T.

Channel Estimation Based on Classification Approaches to Equalization of Time Variant Multipath Channels
,Technical Report of the IEICE,105(29):47-52 2005
Tsutsumi Y.，Tsuda Y. and Shimamura T.

Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
,Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology:2D-09.3 2005
Tsutsumi Y.，Tsuda Y. and Shimamura T.

Formant Frequency Estimation of High-Pitched Speech by Homomorphic Prediction
,Acoustical Science and Technology,26:502-510 2005
Rahman M.S. and Shimamura T.

Frequency Domain Magnitude Banded LMS Algorithm for Equalization of Rapidly Time Variant Channels
,WSEAS Transactions on Electronics,12:1-6 2005
Mohammed H,W, Shimamura T., Cowan C.F.N.

Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:5-8 2005
Kawanaka R., Kobayashi T., Shimamura T., Hosoya T., Takahashi Y.

Non-Stationary Noise Estimation Using Low Frequency Regions for Spectral Subtraction
,IEEE Signal Processing Letters,12:465-468 2005
Yamashita K. and Shimamura T.

Performance Improvement of a Channel Estimation Based Equalizer on Time Variant Multipath Channels
,Proceedings of Signal Processing Symposium:A4-2 2005
Tsuda Y. and Shimamura T.

SIDOモデルを用いたブラインド等化に関する一検討
電子情報通信学会,電子情報通信学会技術研究報告,104719:37-41 2005
藤田昌宏,津田雄亮,島村徹也

Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
,WSEAS Transactions on Signal Processing,3:323-330 2005
Shimamura T.，Yamauchi J.

Time Delay Interpolation by System Response Coefficient Ratios
,IEEE Signal Processing Letters,12:641-644 2005
Gamba J. and Shimamura T.

Variable Step-Size LMS Estimator for Fast Fading Channels
,Technical Report of the IEICE,105:53-58 2005
Oikawa S.，Tsuda Y. and Shimamura T.

White Noise Removal in Image by Iterative Spectral Subtraction Method
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:13-16 2005
Kobayashi T., Shimamura T., Hosoya T., Takahashi Y.

音声強調のための反射係数を利用した雑音パワー推定
,信号処理,9:325-334 2005
緒方伸哉，島村徹也

音声信号のための雑音低減技術　（その１）
,Journal of Signal Processing,9(2):91-98 2005
島村徹也

音声信号のための雑音低減技術　（その２）
,Journal of Signal Processing,9(3):183-188 2005
島村徹也

高性能アクティブノイズキャンセルヘッドフォン開発におけるリアルタイムなディジタル雑音除去の研究
埼玉大学総合研究機構地域共同研究センター産学連携推進部門,埼玉大学地域共同研究センター紀要,6:64-66 2005
島村徹也,和田存功
For the purpose of developing an active noise canceling headphone, techniques of noise reduction are investigated from the viewpoints of digital as well as analogue processing. A prediction based digital method is derived and it is shown that the proposed technique is very useful for noise canceling.
Active noise canceling, Digital processing, Spectral subtraction, Prediction
リポジトリ

洗練フィルタリングを用いたアクティブノイズコントロールシステム
,電子情報通信学会技術研究報告,105(482):45-50 2005
磯崎弘太，津田雄介，島村徹也

反復アルゴリズムを用いたスペクトル引き算法による音声強調
,信号処理,9:255-266 2005
緒方伸哉，島村徹也

音声信号のための順序統計を用いた適応非線形予測器と反復法によるその特性改善
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J87-A(7):899-912 200407
田中啓文,早川晴子,島村徹也
リポジトリ

ピッチ同期加算処理を用いた雑音低減に基づくLPC分析
電子情報通信学会,電子情報通信学会論文誌,J87(A4):458-469 200404
島村徹也、黒岩世進伸

A New Method of Noise Variance Estimation from Low-Order Yule-Walker Equations (Digital Signal Processing)
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E87-A(1):270-274 200401
島村徹也
リポジトリ

A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
IFAC,Proceedings of IFAC Workshop on Adaptation and Learning in Control and Signal Processing:433-438 2004
Mohammed H.W., Shimamura T.

Adaptive Non-linear Prediction of Speech in Impulse Noise
ICA,Proc. 18th International Congress on Acoustics:1675-1678 2004
Ohhashi Y., Tanaka H., Shimamura T.

An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:265-268 2004
Gamba J., Tsuda Y., Shimamura T.

Equalizer-aided time delay tracking based on finite differences
,Proc. 19th IEICE Signal Processing Symposium:B4-4 2004
J. Gamba, Y. Tsuda, and T. Shimamura

Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
ICA,Proc. 18th International Congress on Acoustics:3037-3040 2004
Ogata S., Ebata S., Shimamura T.

Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
,Proc. 47th IEEE International Midwest Symposium on Circuits and Systems:II1 41-144 2004
Shimamura T., Takagi H.

Nonlinear Predictive Analysis of Speech by Iterative Approach
EURASIP,Proc. 12th Europian Signal Processing Conf.:2055-2058 2004
Tanaka H., Shimamura T.

Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:2305-2309 2004
Shimamura T., Yamauchi J.

Pitch Synchronous Addition and Extension for Linear Predictive Analysis of Noisy Speech
EURASIP,Proc. 6th Nordic Signal Processing Symposium:196-199 2004
Shimamura T., Kuroiwa Y.

Reconstruction Filter design for Bone-Conducted Speech
,Proceedings of International Conference on Spoken Language Processing:1-4 2004
Tamiya T., Shimamura T.

反復スペクトル引き算法による雑音重畳画像からの復元
,電子情報通信学会技術研究報告 2004
小林徹也,島村徹也,細谷徹夫,高橋由武

非最小位相通信路における線形トランスバーサル等化器の特性改善 -係数および遅延量の同時適応-
,Proc. 第19回信号処理シンポジウム講演論文集:B4-3 2004
津田雄亮,島村徹也

帯域制限をかけた振幅スペクトルのべき乗に基づく基本周波数抽出法(音声,聴覚)
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J86-A(11):1097-1107 200311
島村徹也,高木浩司
リポジトリ

Noise Estimation Using High Frequency Regions for Spectral Subtraction
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E85-A(3):723-727 200203
島村徹也
リポジトリ

Amplitude Banded RLS Approach to Time Variant Channel Equalization
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E84-A(11):2946-2949 200111
島村徹也
リポジトリ

線形予測分析に基づくホルマント周波数抽出の雑音耐性の改善
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J84-A(6):745-758 200106
島村徹也,趙奇方,高橋淳一,鈴木誠史
リポジトリ

Equalisation of Time Variant Multipath Channels Using Amplitude Banded LMS Algorithms(Digital Signal Processing)(Regular Section)
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E84-A(3):802-812 200103
島村徹也
リポジトリ

平方根及び4乗根パワースペクトルの自己相関に基づくピッチ抽出(研究速報)
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J84-A(3):436-440 200103
島村徹也,吉尾重治,趙奇方,鈴木誠史
リポジトリ

A Fast Converging RLS Equaliser
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E84-A(2):675-680 200102
島村徹也
リポジトリ

ARプレフィルタを用いたIIR型適応等化器とIIR型ウィーナーフィルタ
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J84-A(1):109-112 200101
島村徹也,鈴木誠史
リポジトリ

Weighted Autocorrelation for Pitch Extraction of Noisy Speech
,IEEE Trans. Speech and Audio Processing,9(7):727-730 2001
T. Shimamura and H. Kobayashi

システム同定法を用いた雑音にロバストな音声分析(多次元信号処理とその応用・実現論文小特集)
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J83-A(12):1455-1466 200012
有馬由紀,島村徹也
リポジトリ

An ARMA Prefiltering Approach to Adaptive Equalization
電子情報通信学会,IEICE transactions on fundamentals of electronics, communications and computer sciences,E83-A(10):2035-2039 200010
島村徹也
リポジトリ

対数スペクトルにクリッピングと帯域制限を用いる基本周波数抽出法
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J82-A(7):1115-1122 199907
小林載,島村徹也
リポジトリ

雑音補正による音声のLPC分析の改善
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J81-A(11):1583-1591 199811
島村徹也,趙奇方,鈴木誠史
リポジトリ

対数スペクトルの自己相関関数を用いた搬送波抑圧SSBの離調周波数の推定
電子情報通信学会,電子情報通信学会論文誌. D-II, 情報・システム, II-情報処理,J81-D2(7):1501-1509 199807
金子信一郎,鈴木誠史,島村徹也
リポジトリ

高速スタートアップ等化のためのButler-Cantoni法の適応化
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J81-A(4):622-630 199804
島村徹也,鈴木誠史
リポジトリ

品質劣化音声のためのLPC分析の一改良法
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J80-A(9):1564-1566 199709
島村徹也,國枝伸行,鈴木誠史
リポジトリ

対数スペクトルの自己相関関数を利用したピッチ抽出法
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J80-A(3):435-443 199703
國枝伸行,島村徹也,鈴木誠史
リポジトリ

Equalisation of Time-Variant Communications Channels via Channel Estimation Based Approaches
,Signal Processing,60(2):181-193 1997
T. Shimamura, S.Semani and C.F.N.Cowan

前向き後向き差分関数とフィルタバンクを利用した音声信号の雑音低減
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J79-A(3):541-550 199603
國枝伸行,島村徹也,鈴木誠史
リポジトリ

データ拡張を利用する2次元スペクトル推定法とその改良
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J78-A(8):965-976 199508
島村徹也,繆衛国,鈴木誠史
リポジトリ

ブラインド等化のためのプレフィルタリング
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J78-A(3):323-331 199503
伊藤克子,島村徹也,鈴木誠史
リポジトリ

前向き後向き差分関数による単一正弦波信号の強調
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J77-A(9):1307-1311 199409
國枝伸行,島村徹也,鈴木誠史
リポジトリ

Burg 法のためのデータ予測
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J77-A(8):1182-1185 199408
島村徹也,鈴木誠史
リポジトリ

全極型プレフィルタを用いた IIR 型適応等化器
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J76-A(9):1279-1285 199309
伊藤克子,島村徹也,八嶋弘幸,鈴木誠史
リポジトリ

デルタ変調を利用した分析合成系のピッチ伝送方式(ショートノート)
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J76-A(6):910-912 199306
岩間智史,島村徹也,鈴木誠史
リポジトリ

1次元および2次元信号のためのスペクトルピーク強調法とその応用
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J75-A(12):1783-1791 199212
島村徹也,鈴木誠史
リポジトリ

悪条件下における適応等化のための複素係数を有する最小2乗型適応フィルタ
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J75-A(11):1666-1674 199211
島村徹也,鈴木誠史
リポジトリ

差分関数を利用した音声処理法式 - SPAD -
電子情報通信学会,電子情報通信学会論文誌. A, 基礎・境界,J75-A(11):1769-1772 199211
國枝伸行,島村徹也,鈴木誠史,八嶋弘幸
リポジトリ

Presentation
ポアソン雑音付加画像のための白色化変換を用いた雑音除去
電子情報通信学会,電子情報通信学会東京支部学生研究発表会,185 201103
安部ちかこ, 島村徹也

音声パワｰの無い周波数領域の探索による雑音推定
日本音響学会,日本音響学会2011年春季研究発表会講演論文集,,8-31 201103
赤坂泰司, 島村徹也

Speech Enhancement with a Noise Spectrum Estimation Approach in Nonstationary Environments
日本音響学会,日本音響学会2011年春季研究発表会講演論文集, 国内学会,1-Q-27 201103
A.Saha and T. Shimamura

狭帯域雑音下での基本周波数抽出
日本音響学会,日本音響学会2011年春季研究発表会講演論文集,7-9 201103
茂木沙織, 島村徹也

Noise Estimation Using Only Current Frame of Speech for Spectral Subtraction
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:425-428 201103
A. Matsukawa and T. Shimamura

Noise Reduction Based on Pitch Synchronous Addition and Subtraction for LPC Analysis
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:348-351 201103
L. Liu and T. Shimamura

Performance Improvement of MUSIC by AR Model Based Data Extension
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing:276-279 201103
T. Yokose and T. Shimamura

Extended Fundamental Frequency Extraction Using Exponentiated Amplitude Spectrum with Band-Limitation
IEEE,Proceedings of IEEE International Conference on Signal Acquisition and Processing:V1-168-V1-172 201102
S. Motegi and T. Shimamura

Iterative Edge-Preserving Adaptive Wiener Filter for Image Denoising
IEEE,Proceedings of IEEE International Conference on Signal Acquisition and Processing:V1-168-V1-172 201102
C. Abe and T. Shimamura

双相関､差分､べき乗を利用した雑音混入音声の基本周波数抽出ルゴリズム
電子情報通信学会,電子情報通信学会技術研究報告,SP2010-113:65-69 201101
成田雅俊, 島村徹也

Pitch Determination Using Windowless Autocorrelation Based Cepstrum Method
IEICE,Proceedings of APSIPA Annual Summit and Conference 2010:514-517 201012
M. A. F. M. Rashidul Hasan, M. Shahidur Rahman and T. Shimamura

Autocorrelation and Double Autocorrelation Based線Spectral Representations for Word Recognition in Noisy Environments
電子情報通信学会,信号処理シンポジウム講演論文集,B-10-2:475-478 201011
T. Shimamura and N. D. Nguyen

Convergence Evaluation of a Random Step-Size NLMS Adaptive Algorithm in System Identification
IEEE,Proceedings of IEEE International Conference on Signal Processing:135-138 201010
S. A. Jimaa and T. Shimamura

Noise Estimation Based on Series Expansion of Orthogonal Functions
IEEE,Proceedings of IEEE International Conference on Signal Processing:115-118 201010
T. Akasaka and T. Shimamura

Time-Varying Channel Estimation Using Amplitude-Division Based Parallel NLMS Technique
IEEE,Proceedings of IEEE International Conference on Wireless and Mobile Computing, Networking and Communications:580-585 201010
R. Yasmin and T. Shimamura

Autocorrelation and Double Autocorrelation Representations for a Noisy Word Recognition System
ISCA,Proceedings of INTERSPEECH 2010:1712-1715 201009
T. Shimamura and N. N. Dinh

Pitch Determination Using Autocorrelation Function in Spectral Domain
ISCA,Proceedings of INTERSPEECH 2010:653-656 201009
M. S. Rahman and T. Shimamura

Pitch Characteristics of Bone Conducted Speech
EURASIP,Proceeding of European Signal Processing Conference (EUSIPCO):795-798 201008
M. S. Rahman and T. Shimamura

Accuracy Improvement of Speaker Authentication in Noisy Environments Using Bone-Conducted Speech
IEEE,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:197-200 201008
N. Yamasaki and T. Shimamura

適応等化技術の基礎
電子情報通信学会光通信システム研究会 201007
島村徹也

Toward Speech Communication in Highly Noisy Environments Using Bone Conduction
WSEAS International Conference on Communications 201007
T. Shimamura

Tracking by Nonuniform Amplitude Division Based LMS Algorithm for Time Varying Channels
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:2852-2855 201006
R. Yasmin and T. Shimamura

A Thresholding-Based Image Denoising Method
World Academy of Science, Engineering and Technology,Proceedings of World Academy of Science, Engineering and Technology,65:911-916 201005
S. Suhaila and T. Shimamura

改良重み付き自己相関関数を用いた音声の基本周波数抽出
日本音響学会,日本音響学会2010年春季研究発表会講演論文集,2010春季 201003
成田雅俊

基本周波数抽出のための帯域制限の効果について
日本音響学会,日本音響学会2010年春季研究発表会講演論文集,2010春季 201003
茂木沙織

A Lattice Based Fast Adaptive Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:624-627 201003
K.Tashiro

A New Variable Step Size for Normalized LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:584-587 201003
T.Arnantapunpong

Digital Notch Filter for Noise Reduction in Automatic Train Control System
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:536-539 201003
H.Takekawa

Utilization of Adaptive Line Enhancement and Noise Compensation for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:524-527 201003
C.Tiengwattanatum

Noise Spectrum Estimation Based on Optimum Smoothing for Robust Speech Enhancement
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:293-296 201003
A.Saha

Power Spectrum Estimation Method for Image Denoising by Frequency Domain Wiener Filter
IEEE,Proceedings of IEEE International Conference on Computer and Automation Engineering:608-612 201002
S.Suhaila

音声の2重自己相関関数のスペクトル表現と雑音環境下単語認識システムへの応用
電子情報通信学会,第11回音声言語シンポジウム講演論文集:135-140 200912
グエン・ゴック・ディン

直交関数系の級数展開に基づくスペクトル引き算
電子情報通信学会,第11回音声言語シンポジウム講演論文集:129-134 200912
赤坂泰司

Convergence Evaluation of Variable Step-Size NLMS Algorithms in Adaptive Channel Equalization
IEEE,Proceedings of IEEE International Symposium on Signal Processing and Information Technology:145-150 200912
S.A.Jimaa

Reverberated Speech Enhancement Using Neural Networks
IEEE,Proceedings of IEEE Intelligent Signal Processing and Communication Systems:441-444 200912
B.D.Dufera

ラティスフィルタを用いた高速適応アルゴリズム
電子情報通信学会,電子情報通信学会技術研究報告,SIP2009-59 200910
田代和義

画像復元のための反復的エッジ保存適応ウィナーフィルタ
電子情報通信学会,電子情報通信学会技術研究報告,SIP2009-60 200910
安部ちかこ

並列型適応フィルタを用いた音響エコーキャンセラの提案
日本音響学会,日本音響学会2009年秋季研究発表会講演論文集,2009秋季 200909
清水太治郎

基本周波数抽出のためのスペクトル引き算の効果について
日本音響学会,日本音響学会2009年秋季研究発表会講演論文集,2009秋季 200909
茂木沙織

振幅スペクトルのべき乗引き算に基づく音声の基本周波数抽出
電子情報通信学会,電子情報通信学会技術研究報告,SP2009-39 200906
茂木沙織

Equalization of Time-Variant Communications Channels via Adaptive AR Prefiltering
WRI,Proceedings of WRI World Congress on Computer Science and Information Engineering:327-331 200903
Shimamura T.

スペクトル引き算を利用したウィナーフィルタによる画像復元
電子情報通信学会,電子情報通信学会2009総合大会講演論文集:A-4-8 200903
江田慎太郎

改良雑音スペクトル推定を用いたウィナーフィルタリングによる画像復元
電子情報通信学会,電子情報通信学会2009総合大会講演論文集:A-4-9 200903
古屋拡子

反復適応ウィナーフィルタを用いた画像復元
電子情報通信学会,電子情報通信学会2009総合大会講演論文集:A-4-7 200903
安部ちかこ

Dual Adaptive Pre-Whitening Filters for LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:1-4 200903
Tashiro K.

Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:165-168 200903
Nakamura N.

Image Restoration via Wiener Filtering with Improved Noise Estimation
WSEAS,Proceedings of WSEAS International Conference on Signal Processing, Robotics and Automation:315-320 200902
Furuya H.

Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization
IEEE,Proceedings of International Conference on Computer and Information Technology:768-772 200812
Khan, M.L.R.

High Pitch Source Isolation Using Complex Cepstrum in the Autocorrelation Domain
IEEE,Proceedings of IEEE Asia Pacific Conference on Circuits and Systems:1284-1287 200812
Derebssa B

双対の適応白色化フィルタを用いたLMSアルゴリズム
電子情報通信学会,信号処理シンポジウム講演論文集:195-200 200811
田代和義

線形予測誤差を用いた骨導音声の品質改善
電子情報通信学会,信号処理シンポジウム講演論文集:164-169 200811
杉山貴紀

An Efficient and Effective Variable Step Size NLMS Algorithm
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:1640-1643 200810
Takekawa H.

Adaptive Line Enhancer for Time Delay Estimation of Ultrasonic Echoes
IEEE,Proceedings of International Symposium on Communication and Information Technology:368-371 200810
Tiengwattanatum C.

Amplitude-Division Parallel LMS Estimator
IEEE,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:950-953 200808
Shimamura T.

Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm
IEEE,Proceedings of IEEE International Networking and Communications Conference:23-26 200805
Takekawa H.

高騒音環境下における骨導音声を用いた話者認識
日本音響学会,日本音響学会春期研究発表会講演論文集:3-11-7 200803
飯島昌平

デュアルマイクロホンでの時間遅延を利用した音声強調手法
日本音響学会,日本音響学会春期研究発表会講演論文集:1-P-16 200803
池田達也

縦続型適応非線形予測器を用いた音声信号の予測分析
日本音響学会,日本音響学会春期研究発表会講演論文集:1-11-9 200803
山崎直人

Iterative Cross-Correlation Method for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:387-390 200803
Nakamura N.

Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:291-294 200803
Sugiyama T.

Bone-Conducted Speech for Speaker Verification
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:172-175 200803
Iijima S.

Wavelet Based Denoising for Images Degraded by Poisson Noise
IASTED,Proceedings of IASTED International Conference on Biomedical Engineering:436-440 200802
Shimamura T., Eda S., Ito T., Kuwano Y. and Takahashi Y.

Amplitude Banded Sato Algorithm for Blind Channel Equalization
IEEE,Proceedings of IEEE International Conference on Signal Processing and Communications:1463-1466 200711
Khan M.L.R.

Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
NOLTA,Proceedings of International Symposium on Nonlinear Theory and Its Applications:441-444 200709
Eda S.

Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
IASTED,Proceedings of IASTED International Conference on Signal and Image Processing:576-205 200708
Shimamura T.

Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
IEEE,Proceedings of IEEE International Workshop on Nonlinear Dynamics of Electronic Systems:289-292 200707
Shimamura T.

Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
NICOGRAPH,Proceedings of NICOGRAPH International 2007 200705
Xin W.

ディジタル無線通信における雑音にロバストな検波のための一手法
電子情報通信学会,電子情報通信学会総合大会講演論文集:B-5-83 200703
飯島昌平

Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:583-586 200703
Kamata K.

Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:213-216 200703
Tiengwattanatum C.

Indirect Cross-Correlation Method for Time Delay Estimation
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:74-77 200703
Nakamura N.

Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
WSEAS,Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications:125-129 200701
Tanaka H.

Discrete Cosine Transform Domain Parallel LMS Equalizer
WSEAS,Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications:119-124 200701
Mohammed H.W.

A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels
IEEE,Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems:845-848 200612
Oikawa S.

Adaptive Non-Linear Prediction with Order Statistics for Speech Signals in Mixture Noise
IEEE,Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems:295-298 200612
Tanaka H.

Active Noise Control Using A Refined Filtering Approach
ICA,Proceedings of the 35th International Congress and Exposition on Noise Control Engineering 200612
Isozaki K.

Wavelet Based Keyframe Extraction Method from Motion Capture Data
ADADA,Proceedings on Asia Digital Art and Design Association:128-129 200612
Kondo K.

Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
IEEE,Proceedings of International Conference on Signals and Electronic Systems 200609
Rahman M.S.

Learning for Bone-Conducted Speech via Adaptive and Neural Filters
IEEE,Proceedings of International Conference on Signals and Electronic Systems 200609
Rahman M.S.

Pitch Determination using Aligned AMDF
ISCA,Proceedings of International Conference on Spoken Language Processing:1714-1717 200609
Rahman M.S.

A Harsh Noise Assessment Measure for Speech Enhancement
EURASIP,Proceedings of European Conference on Signal Processing 200609
Yamashita K.

Improving Bone-Conducted Speech Quality via Neural Network
IEEE,Proceedings of IEEE International Symposium on Signal Processing and Information Technology:628-632 200608
Shimamura T.

Reconstruction Filter design for Bone-Conducted Speech
ESCA,Proceedings of International Conference on Spoken Language Processing:1-4 200604
Tamiya T.

SIDOモデルを用いたブラインド等化に関する一検討
電子情報通信学会,電子情報通信学会技術研究報告,104719:37-41 200604
藤田昌宏

反復スペクトル引き算法による雑音重畳画像からの復元
電子情報通信学会,電子情報通信学会技術研究報告 200604
小林徹也

骨導音声の品質改善について　（その１）
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:243-244 200603
田宮俊樹

反復RFアルゴリズムを利用したリアルタイム騒音制御
日本音響学会,日本音響学会春季研究発表会講演論文集,3-Q-16:659-660 200603
大橋祐一郎

縦列型適応フィルタを用いたアクティブノイズコントロールシステムの提案
日本音響学会,日本音響学会春季研究発表会講演論文集,2-9-8:769-770 200603
磯崎弘太

スペクトル引き算のための雑音の煩雑さを考慮した雑音評価法
日本音響学会,日本音響学会春季研究発表会講演論文集,1-Q-4:341-342 200603
山下浩平

線形予測フィルタを用いた適応音声強調
日本音響学会,日本音響学会春季研究発表会講演論文集,1-Q-29:391-392 200603
田中啓文

騒音環境下での適応フィルタによる骨導音声の品質改善
日本音響学会,日本音響学会春季研究発表会講演論文集,1-Q-31:395-396 200603
田宮俊樹

A Study on Normalized LMS Algorithm Using Refined Filtering Technique
WSEAS,Proceedings of 5th WSEAS International Conference on Signal Processing, Robotics and Automation:264-267 200602
Tsuda Y.

Active Noise Control Using Cascaded Adaptive Filters
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Isozaki K.

Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Wondimagegnehu M.H.

Noise Estimation Using Multifrequency Regions for Spectral Subtraction
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing 2006
Yamashita K.

A Parallel Estimator with LMS Adaptation for Fast Fading Channels
IEEE,Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology:2C-12.4 200511
Oikawa S.

Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
IEEE,Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology:2D-09.3 200511
Tsutsumi Y.

Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
WSEAS,Proceedings of WSEAS International Conference on Electronics, Control and Signal Processing:47-52 200511
Shimamura T.

Speech Enhancement Using a Technique of Adaptive Bias Suppression
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:540-544 200510
Tanaka H.

Voice Source Modeling for Accurate Speech Analysis
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:305-309 200510
Shimamura T.

Quality Improvement of Bone-Conducted Speech
IEEE,Proceedings of European Conference on Circuit Theory and Design 200508
Shimamura T.

A Reconstruction Filter for Bone-Conducted Speech
IEEE,Proceedings of IEEE International Midwest Symposium on Circuits and Systems:1847-1850 200508
Shimamura T.

Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:6288-6291 200505
Shimamura T.

Linear Prediction Using Homomorphic Deconvolution in the Autocorrelation Domain
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:2855-2858 200505
Rahman M.S.

Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
IEEE, EURASIP,Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing:404-409 200505
Tanaka H.

Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
IEEE, EURASIP,Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing:6-9 200505
Shimamura T.

LMS-VSLMS縦列接続による適応非線形予測分析
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:259-260 200503
田中啓文

雑音スペクトルの多重処理を用いた改良スペクトル引き算法による音声強調
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:255-256 200503
山下浩平

白色雑音とインパルス雑音の混合環境下における音声信号の適応非線形予測分析
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:253-254 200503
大橋祐一郎

アクティブノイズコントロールへの改良正規化LMSアルゴリズムの適用
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:421-422 200503
磯崎弘太

骨導音声の品質改善について　（その２）
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:239-240 200503
間宮淳一郎

適応バイアス抑制技術を用いた音声強調
日本音響学会,日本音響学会2005年春季研究発表会講演論文集:237-238 200503
山村尚己

Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:5-8 2005
Kawanaka R.

White Noise Removal in Image by Iterative Spectral Subtraction Method
RISP,Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing:13-16 2005
Kobayashi T.

A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
IFAC,Proceedings of IFAC Workshop on Adaptation and Learning in Control and Signal Processing:433-438 2004
Mohammed H.W.

Adaptive Non-linear Prediction of Speech in Impulse Noise
ICA,Proc. 18th International Congress on Acoustics:1675-1678 2004
Ohhashi Y.

An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
IEEE,Proceedings of IEEE International Symposium on Circuits and Systems:265-268 2004
Gamba J.

Equalizer-aided time delay tracking based on finite differences
電子情報通信学会,Proc. 19th IEICE Signal Processing Symposium 2004
J. Gamba

Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
ICA,Proc. 18th International Congress on Acoustics:3037-3040 2004
Ogata S.

Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
IEEE,Proc. 47th IEEE International Midwest Symposium on Circuits and Systems:II 141-144 2004
Shimamura T.

Nonlinear Predictive Analysis of Speech by Iterative Approach
EURASIP,Proc. 12th Europian Signal Processing Conf.:2055-2058 2004
Tanaka H.

Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
IEEE,Proceedings of Asilomar Conference on Signals, Systems and Computers:2305-2309 2004
Shimamura T.

Pitch Synchronous Addition and Extension for Linear Predictive Analysis of Noisy Speech
EURASIP,Proc. 6th Nordic Signal Processing Symposium:196-199 2004
Shimamura T.

非最小位相通信路における線形トランスバーサル等化器の特性改善 -係数および遅延量の同時適応-
電子情報通信学会,Proc. 第19回信号処理シンポジウム講演論文集 2004
津田雄亮