Equalization of Time-Variant Communications Channels via Adaptive AR Prefiltering Shimamura T
First page：327,
Last page：331, 2009
DOI：https://doi.org/10.1109/CSIE.2009.250DOI ID：10.1109/CSIE.2009.250 Dual Adaptive Pre-Whitening Filters for LMS Algorithm
Tashiro K; Shimamura T
First page：1, Last page：4, 2009
Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
Nakamura N; Ikeda T; Shimamura T
First page：165, Last page：168, 2009
Image Restoration via Wiener Filtering with Improved Noise Estimation
Furuya H; Eda S; Shimamura T
First page：315, Last page：320, 2009
Image Restoration via Wiener Filtering in the Frequency Domain
Furuya H; Eda S; Shimamura T
Volume：5, Number：Issue 2, First page：63, Last page：73, 2009
Equalization of time-variant communications channels via adaptive AR prefiltering Tetsuya Shimamura
2009 WRI World Congress on Computer Science and Information Engineering, CSIE 2009,
Volume：6,
First page：327,
Last page：331, 2009
For the purpose of time-variant channel equalization, a novel structure of equalizer is proposed. An autoregressive (AR) filter is used as the prefilter and a finite impulse response (FIR) filter is cascaded. The AR prefilter is constructed from channel estimation results adaptively, and the cascaded FIR filter behaves as an adaptive FIR equalizer. It is shown that the resulting infinite impulse response (IIR) equalizer is closely related with the IIR-type Wiener filter. Simulations demonstrate that the proposed equalizer provides better performance than the FIR equalizer with the least mean square adaptation in time-variant environments. © 2008 IEEE.
English
DOI：https://doi.org/10.1109/CSIE.2009.250DOI ID：10.1109/CSIE.2009.250,
SCOPUS ID：71049116017 Dual Adaptive Pre-Whitening Filters for LMS Algorithm
Tashiro K; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：1, Last page：4, 2009
Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
Nakamura N; Ikeda T; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：165, Last page：168, 2009
Image Restoration via Wiener Filtering with Improved Noise Estimation
Furuya H; Eda S; Shimamura T
Proceedings of WSEAS International Conference on Signal Processing, Robotics and Automation, First page：315, Last page：320, 2009
Image Restoration via Wiener Filtering in the Frequency Domain
Furuya H; Eda S; Shimamura T
WSEAS Transactions on Signal Processing, Volume：5, Number：Issue 2, First page：63, Last page：73, 2009
Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization Khan, M.L.R; Wondimagegnehu M.H; Shimamura T
First page：768,
Last page：772, 2008
DOI：https://doi.org/10.1109/ICCITECHN.2008.4803015DOI ID：10.1109/ICCITECHN.2008.4803015 Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm Jimaa S; Shimamura T; Takekawa H
First page：23,
Last page：26, 2008
DOI：https://doi.org/10.1109/INCC.2008.4562685DOI ID：10.1109/INCC.2008.4562685 Iterative Cross-Correlation Method for Time Delay Estimation
Nakamura N; Shimamura T
First page：387, Last page：390, 2008
Wavelet Based Denoising for Images Degraded by Poisson Noise
Shimamura T; Eda S; Ito T; Kuwano Y; Takahashi Y
First page：436, Last page：440, 2008
騒音環境下に有効な骨導・気導一体型マイクロホン
島村徹也
First page：11, Last page：16, 2008
双対の適応白色化フィルタを用いたLMSアルゴリズム
田代和義; 島村徹也
First page：195, Last page：200, 2008
線形予測誤差を用いた骨導音声の品質改善
杉山貴紀; 島村徹也; 八嶋弘幸
First page：164, Last page：169, 2008
Special Section on Papers Awarded the Student Paper Award at NCSP'08 Editor's Note
Shimamura T
Volume：12, Number：4, First page：269, Last page：270, 2008
Special Section on Nonlinear Circuits and Signal Processing Editor's Note
Shimamura T
Volume：12, Number：6, First page：412, Last page：413, 2008
Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization Khan, M.L.R; Wondimagegnehu M.H; Shimamura T
Proceedings of International Conference on Computer and Information Technology,
First page：768,
Last page：772, 2008
DOI：https://doi.org/10.1109/ICCITECHN.2008.4803015DOI ID：10.1109/ICCITECHN.2008.4803015 High Pitch Source Isolation Using Complex Cepstrum in the Autocorrelation Domain Bisrat Derebssa; Tetsuya Shimamura
2008 IEEE ASIA PACIFIC CONFERENCE ON CIRCUITS AND SYSTEMS (APCCAS 2008), VOLS 1-4,
First page：1284,
Last page：1287, 2008
This paper proposes a technique for improving the performance of linear predictive analysis by using homomorphic deconvolution. Estimating the vocal tract transfer function using linear prediction is often inaccurate when the speaker speaks with a high pitch. This is due to the convolution of the glottal output signal, which is a periodic impulse, with the vocal tract impulse response. We propose a method that will remove this convolution by using complex cepstrum homomorphic deconvolution in the autocorrelation domain. The technique has been tested for synthetic and natural speech of varying pitch frequency and has improved performance on formant frequency estimation than a similar technique proposed recently.
IEEE, English
DOI：https://doi.org/10.1109/APCCAS.2008.4746262DOI ID：10.1109/APCCAS.2008.4746262,
Web of Science ID：WOS:000268007100317 An Efficient and Effective Variable Step Size NLMS Algorithm Takekawa H; Shimamura T; Jimaa S
Proceedings of Asilomar Conference on Signals, Systems and Computers,
First page：1640,
Last page：1643, 2008
DOI：https://doi.org/10.1109/ACSSC.2008.5074702DOI ID：10.1109/ACSSC.2008.5074702 Adaptive Line Enhancer for Time Delay Estimation of Ultrasonic Echoes Chayanin Tiengwattanatum; Nakamura Naoyuki; Tetsuya Shimamura
2008 INTERNATIONAL SYMPOSIUM ON COMMUNICATIONS AND INFORMATION TECHNOLOGIES,
First page：368,
Last page：371, 2008
A new method of time delay estimation for ultrasonic echoes in noisy environment is proposed. By using the least mean square algorithm for adaptive line enhancer as a pre-filter and following by a cross correlator can achieve better performance than the conventional method. Computer simulations results confirm the superiority of the proposed method even at very low signal-to-noise ratios.
IEEE, English
DOI：https://doi.org/10.1109/ISCIT.2008.4700215DOI ID：10.1109/ISCIT.2008.4700215,
Web of Science ID：WOS:000265524600073 Amplitude-Division Parallel LMS Estimator Tetsuya Shimamura; Shingo Oikawa; Yusuke Tsuda
2008 51ST MIDWEST SYMPOSIUM ON CIRCUITS AND SYSTEMS, VOLS 1 AND 2,
First page：950,
Last page：953, 2008
This paper presents a novel tracking technique for rapidly time-varying channels. The proposed scheme supposes multiple linear transversal filters (estimators), which are constructed in a parallel structure. The coefficient vectors for each estimator are adapted by the least mean square (LMS) algorithm according to the information of the channel coefficient values. Computer simulation results show that the proposed estimator provides an improvement relative to the conventional prediction based LMS estimators in various fade rate conditions.
IEEE, English
DOI：https://doi.org/10.1109/MWSCAS.2008.4616958DOI ID：10.1109/MWSCAS.2008.4616958,
ISSN：1548-3746,
Web of Science ID：WOS:000261729500238 Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm Jimaa S; Shimamura T; Takekawa H
Proceedings of IEEE International Networking and Communications Conference,
First page：23,
Last page：26, 2008
DOI：https://doi.org/10.1109/INCC.2008.4562685DOI ID：10.1109/INCC.2008.4562685 Iterative Cross-Correlation Method for Time Delay Estimation
Nakamura N; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：387, Last page：390, 2008
Wavelet Based Denoising for Images Degraded by Poisson Noise
Shimamura T; Eda S; Ito T; Kuwano Y; Takahashi Y
Proceedings of IASTED International Conference on Biomedical Engineering, First page：436, Last page：440, 2008
騒音環境下に有効な骨導・気導一体型マイクロホン
島村徹也
ナノ材料・IT新技術説明会資料集, First page：11, Last page：16, 2008
双対の適応白色化フィルタを用いたLMSアルゴリズム
田代和義; 島村徹也
信号処理シンポジウム講演論文集, First page：195, Last page：200, 2008
線形予測誤差を用いた骨導音声の品質改善
杉山貴紀; 島村徹也; 八嶋弘幸
信号処理シンポジウム講演論文集, First page：164, Last page：169, 2008
Special Section on Papers Awarded the Student Paper Award at NCSP'08 Editor's Note
Shimamura T
Journal of Signal Processing, Volume：12, Number：4, First page：269, Last page：270, 2008
Special Section on Nonlinear Circuits and Signal Processing Editor's Note
Shimamura T
Journal of Signal Processing, Volume：12, Number：6, First page：412, Last page：413, 2008
Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
Eda S; Shimamura T
First page：441, Last page：444, 2007
Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
Shimamura T
First page：576, Last page：205, 2007
Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
Shimamura T
First page：289, Last page：292, 2007
Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
Tanaka H; Ohhashi Y; Shimamura T
First page：125, Last page：129, 2007
Discrete Cosine Transform Domain Parallel LMS Equalizer
Mohammed H.W; Shimamura T
First page：119, Last page：124, 2007
癒し音楽における1/fゆらぎと高周波成分との関連性
島村徹也; 小花あゆみ
Volume：26, First page：25, Last page：30, 2007
音声信号の非線形予測と信号表現に関する研究
島村徹也
Volume：5 (18年度), First page：602, Last page：603, 2007
効率的な時間遅延推定のための間接的差分関数法
中村尚之; 島村徹也
First page：401, Last page：405, 2007
可変ステップサイズ正規化LMSアルゴリズムの一提案
竹川英樹; 島村徹也
First page：466, Last page：471, 2007
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
2007
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
Volume：6, First page：11, Last page：18, 2007
Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
Tiengwattanatum C; Mohammed H.W; Shimamura T
First page：213, Last page：216, 2007
Performance of Adaptive Nonlinear Predictor with Order Statistics in Impulsive Noise
Tanaka H; Ohhashi Y; Shimamura T
Volume：2, 2007
Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
Sugiyama T; Shimamura T; Yashima H
First page：291, Last page：294, 2007
Indirect Cross-Correlation Method for Time Delay Estimation
Nakamura N; Shimamura T
First page：74, Last page：77, 2007
Complementary and Phase Banded LMS Equalizers for Rapidly Time-Varying Channels
Mohammed H.W; Shimamura T
Volume：11, First page：51, Last page：60, 2007
A Parallel Equalizer with LMS Adaptation in Discrete Cosine Transform Domain
Mohammed H.W; Shimamura T
Volume：2, 2007
Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
Kamata K; Yamashita K; Shimamura T; Furukawa T
First page：583, Last page：586, 2007
Bone-Conducted Speech for Speaker Verification
Iijima S; Shimamura T
First page：172, Last page：175, 2007
AMPLITUDE BANDED SATO ALGORITHM FOR BLIND CHANNEL EQUALIZATION Muhammad Lutfor Rahman Khan; Mohammed H. Wondimagegnehu; Tetsuya Shimamura
ICSPC: 2007 IEEE INTERNATIONAL CONFERENCE ON SIGNAL PROCESSING AND COMMUNICATIONS, VOLS 1-3, PROCEEDINGS,
First page：1463,
Last page：1466, 2007
In this paper we propose a novel non-linear blind adaptive algorithm called the Amplitude Banded Sato (ABSato) algorithm for equalization of communication channels. The ABSato algorithm is derived as a modified version of the Amplitude Banded Least Mean Square (ABLMS) algorithm addressed by Shimamura et al. recently The capability of nonlinear classification the ABLMS algorithm inherently possesses is kept in the ABSato algorithm, resulting in an improvement of equalization performance. Mean square error as well as bit error rate performances are investigated on simple communication channel models. Observation of simulation results show that the ABSato algorithm provides better performance than the standard Sato algorithm on all the communication channel models.
IEEE, English
DOI：https://doi.org/10.1109/ICSPC.2007.4728606DOI ID：10.1109/ICSPC.2007.4728606,
Web of Science ID：WOS:000266406700367 Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
Eda S; Shimamura T
Proceedings of International Symposium on Nonlinear Theory and Its Applications, First page：441, Last page：444, 2007
Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
Shimamura T
Proceedings of IASTED International Conference on Signal and Image Processing, First page：576, Last page：205, 2007
Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
Shimamura T
Proceedings of IEEE International Workshop on Nonlinear Dynamics of Electronic Systems, First page：289, Last page：292, 2007
Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
Tanaka H; Ohhashi Y; Shimamura T
Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications, First page：125, Last page：129, 2007
Discrete Cosine Transform Domain Parallel LMS Equalizer
Mohammed H.W; Shimamura T
Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications, First page：119, Last page：124, 2007
癒し音楽における1/fゆらぎと高周波成分との関連性
島村徹也; 小花あゆみ
音楽音響研究会資料, Volume：26, First page：25, Last page：30, 2007
音声信号の非線形予測と信号表現に関する研究
島村徹也
総合研究機構研究プロジェクト研究成果報告書, Volume：5 (18年度), First page：602, Last page：603, 2007
効率的な時間遅延推定のための間接的差分関数法
中村尚之; 島村徹也
信号処理シンポジウム講演論文集, First page：401, Last page：405, 2007
可変ステップサイズ正規化LMSアルゴリズムの一提案
竹川英樹; 島村徹也
信号処理シンポジウム講演論文集, First page：466, Last page：471, 2007
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
Proceedings of NICOGRAPH, 2007
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
International Journal of Asia Digital Art and Design, Volume：6, First page：11, Last page：18, 2007
Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
Tiengwattanatum C; Mohammed H.W; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：213, Last page：216, 2007
Performance of Adaptive Nonlinear Predictor with Order Statistics in Impulsive Noise
Tanaka H; Ohhashi Y; Shimamura T
WSEAS Transactions on Signal Processing, Volume：2, 2007
Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
Sugiyama T; Shimamura T; Yashima H
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：291, Last page：294, 2007
Linear Prediction Using Refined Autocorrelation Function M. Shahidur Rahman; Tetsuya Shimamura
EURASIP JOURNAL ON AUDIO SPEECH AND MUSIC PROCESSING,
Volume：2007,
First page：Article ID 45962, 9 Pages, 2007
This paper proposes a new technique for improving the performance of linear prediction analysis by utilizing a refined version of the autocorrelation function. Problems in analyzing voiced speech using linear prediction occur often due to the harmonic structure of the excitation source, which causes the autocorrelation function to be an aliased version of that of the vocal tract impulse response. To estimate the vocal tract characteristics accurately, however, the effect of aliasing must be eliminated. In this paper, we employ homomorphic deconvolution technique in the autocorrelation domain to eliminate the aliasing effect occurred due to periodicity. The resulted autocorrelation function of the vocal tract impulse response is found to produce significant improvement in estimating formant frequencies. The accuracy of formant estimation is verified on synthetic vowels for a wide range of pitch frequencies typical for male and female speakers. The validity of the proposed method is also illustrated by inspecting the spectral envelopes of natural speech spoken by high-pitched female speaker. The synthesis filter obtained by the current method is guaranteed to be stable, which makes the method superior to many of its alternatives. Copyright (C) 2007 M. S. Rahman and T. Shimamura.
SPRINGER INTERNATIONAL PUBLISHING AG, English
DOI：https://doi.org/10.1155/2007/45962DOI ID：10.1155/2007/45962,
ISSN：1687-4722,
Web of Science ID：WOS:000207767400001 Indirect Cross-Correlation Method for Time Delay Estimation
Nakamura N; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：74, Last page：77, 2007
Complementary and Phase Banded LMS Equalizers for Rapidly Time-Varying Channels
Mohammed H.W; Shimamura T
Journal of Signal Processing, Volume：11, First page：51, Last page：60, 2007
A Parallel Equalizer with LMS Adaptation in Discrete Cosine Transform Domain
Mohammed H.W; Shimamura T
WSEAS Transactions on Signal Processing, Volume：2, 2007
Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
Kamata K; Yamashita K; Shimamura T; Furukawa T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：583, Last page：586, 2007
Bone-Conducted Speech for Speaker Verification
Iijima S; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：172, Last page：175, 2007
Adaptive non-linear prediction for speech signals in mixture noise environments Hirobumi Tanaka; Yuichiroh Ohhashi; Tetsuya Shimamura
2006 International Symposium on Intelligent Signal Processing and Communications, ISPACS'06,
First page：295,
Last page：298, 2007
For robust prediction analysis for speech signals in impulsive noise environments, we had proposed the OSLMS with AS predictor in [1]. Assuming general noise environments, however, we can not neglect background noises. Therefore, in this paper, we studied properties of the LPC in a mixture noise environment which has a content of impulsive noises and white Gaussian noises, and prove that an adaptive predictor perform more effectively in the mixture noises which has white noises with low power than batch predictors. Furthermore, we conducted prediction experiments using 3 adaptive predictors including the OSLMS with AS in the mixture noise environments. As a result, the OSLMS with AS predictor provided good performance in mixture noise environments which has practical white noise power, too. © 2006 IEEE.
English
DOI：https://doi.org/10.1109/ISPACS.2006.364890DOI ID：10.1109/ISPACS.2006.364890,
SCOPUS ID：45249095876 A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels Oikawa S; Tsuda Y; Shimamura T
First page：845,
Last page：848, 2006
DOI：https://doi.org/10.1109/ISPACS.2006.364777DOI ID：10.1109/ISPACS.2006.364777 Active Noise Control Using A Refined Filtering Approach
Isozaki K; Tsuda Y; Shimamura T
2006
Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
First page：128, Last page：129, 2006
Learning for Bone-Conducted Speech via Adaptive and Neural Filters
Shimamura T; Tamiya T
2006
A Harsh Noise Assessment Measure for Speech Enhancement
Yamashita K; Shimamura T
2006
Speech Analysis Based on Modeling the Effective Voice Source(Speech Analysis, Statistical Modeling for Speech Processing) 島村徹也
Volume：E89-D,
Number：3,
First page：1107,
Last page：1115, 2006
DOI：https://doi.org/10.1093/ietisy/e89-d.3.1107DOI ID：10.1093/ietisy/e89-d.3.1107,
ISSN：0916-8532,
CiNii Articles ID：110004719387 A Study on Normalized LMS Algorithm Using Refined Filtering Technique
Tsuda Y; Shimamura T
First page：264, Last page：267, 2006
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
島村徹也; 和田存功
Volume：7, First page：64, Last page：64, 2006
Noise Estimation Using Multifrequency Regions for Spectral Subtraction
Yamashita T; Shimamura T
Volume：10, First page：275, Last page：278, 2006
Noise Estimation Using Multifrequency Regions for Spectral Subtraction
Yamashita K; Shimamura T
2006
Refined Filtering for Normalized LMS Algorithm
Tsuda Y; Shimamura T
Volume：2, First page：261, Last page：264, 2006
A Refined Filtering Approach to Adaptive Line Enhancement
Tsuda Y; Shimamura T
First page：141, Last page：146, 2006
Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
Rahman M.S; Tanaka H; Shimamura T
2006
Adaptive Time Variant Channel Equalization Using Phase Banded LMS Algorithm
Mohammed H.W; Shimamura T
Volume：10, First page：227, Last page：230, 2006
Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
Mohammed H. W; Shimamura T
2006
Active Noise Control Using Cascaded Adaptive Filters
Isozaki K; Tsuda Y; Shimamura T
Volume：10, First page：279, Last page：282, 2006
Active Noise Control Using Cascaded Adaptive Filters
Isozaki K; Tsuda Y; Shimamura T
2006
A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels Oikawa S; Tsuda Y; Shimamura T
Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems,
First page：845,
Last page：848, 2006
DOI：https://doi.org/10.1109/ISPACS.2006.364777DOI ID：10.1109/ISPACS.2006.364777 Active Noise Control Using A Refined Filtering Approach
Isozaki K; Tsuda Y; Shimamura T
Proceedings of the 35th International Congress and Exposition on Noise Control Engineering, 2006
Wavelet Based Keyframe Extraction Method from Motion Capture Data
Xin W; Kondo K; Tateno K; Konma T; Shimamura T
Proceedings on Asia Digital Art and Design Association, First page：128, Last page：129, 2006
Learning for Bone-Conducted Speech via Adaptive and Neural Filters
Shimamura T; Tamiya T
Proceedings of International Conference on Signals and Electronic Systems, 2006
A Harsh Noise Assessment Measure for Speech Enhancement
Yamashita K; Shimamura T
Proceedings of European Conference on Signal Processing, 2006
Improving Bone-Conducted Speech Quality via Neural Network Shimamura T; Mamiya J; Tamiya T
Proceedings of IEEE International Symposium on Signal Processing and Information Technology,
First page：628,
Last page：632, 2006
DOI：https://doi.org/10.1109/ISSPIT.2006.270876DOI ID：10.1109/ISSPIT.2006.270876 Speech Analysis Based on Modeling the Effective Voice Source(Speech Analysis, Statistical Modeling for Speech Processing) 島村徹也
IEICE transactions on information and systems,
Volume：E89-D,
Number：3,
First page：1107,
Last page：1115, 2006
DOI：https://doi.org/10.1093/ietisy/e89-d.3.1107DOI ID：10.1093/ietisy/e89-d.3.1107,
ISSN：0916-8532,
CiNii Articles ID：110004719387 A Study on Normalized LMS Algorithm Using Refined Filtering Technique
Tsuda Y; Shimamura T
Proceedings of 5th WSEAS International Conference on Signal Processing, Robotics and Automation, First page：264, Last page：267, 2006
Coefficients - Delay simultaneous adaptation scheme for linear equalization of nonminimum phase channels Y Tsuda; J Gamba; T Shimamura
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES,
Volume：E89A,
Number：1,
First page：248,
Last page：259, Jan. 2006
An efficient adaptation technique of the delay is introduced for accomplishing more accurate adaptive linear equalization of nonminimum phase channels. It is focused that the filter structure and adaptation procedure of the adaptive Butler-Cantoni (ABC) equalizer is very suitable to deal with a variable delay for each iteration, compared with a classical adaptive linear transversal equalizer (LTE). We derive a cost function by comparing the system mismatch of an optimum equalizer coefficient vector with an equalizer coefficient vector with several delay settings. The cost function is square of difference of absolute values of the first element and the last element for the equalizer coefficient vector. The delay adaptation method based on the cost function is developed, which is involved with the ABC equalizer. The delay is adapted by checking the first and last elements of the equalizer coefficient vector and this results in an LTE providing a lower mean square error level than the other LTEs with the same order. We confirm the performance of the ABC equalizer with the delay adaptation method through computer simulations.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English
DOI：https://doi.org/10.1093/ietfec/e89-a.1.248DOI ID：10.1093/ietfec/e89-a.1.248,
ISSN：0916-8508,
eISSN：1745-1337,
CiNii Articles ID：110003486133,
Web of Science ID：WOS:000234948500035 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島村徹也; 和田存功
埼玉大学地域共同研究センター紀要,
Volume：7,
First page：64,
Last page：64, 2006
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.
Japanese
ISSN：1347-4758,
CiNii Articles ID：120001371317 Noise Estimation Using Multifrequency Regions for Spectral Subtraction
Yamashita T; Shimamura T
Journal of Signal Processing, Volume：10, First page：275, Last page：278, 2006
Noise Estimation Using Multifrequency Regions for Spectral Subtraction
Yamashita K; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Refined Filtering for Normalized LMS Algorithm
Tsuda Y; Shimamura T
WSEAS Transactions on Signal Processing, Volume：2, First page：261, Last page：264, 2006
A Refined Filtering Approach to Adaptive Line Enhancement
Tsuda Y; Shimamura T
回路とシステム軽井沢ワークショップ講演論文集, First page：141, Last page：146, 2006
Pitch Determination Using Aligned AMDF
M. Shahidur Rahman; Hirobumi Tanaka; Tetsuya Shimamura
INTERSPEECH 2006 AND 9TH INTERNATIONAL CONFERENCE ON SPOKEN LANGUAGE PROCESSING, VOLS 1-5, First page：1714, Last page：1717, 2006
A pitch determination method based on AMDF (Average Magnitude Difference Function) is proposed in this paper. The AMDF is often used to determine the pitch parameter in real-time speech processing applications. Failing trend of AMDF at higher lags, however, makes the method vulnerable to octave errors (pitch doubling or halving). In this paper, we propose an alignment technique that effectively eliminates the falling trend by aligning the AMDF peaks along a straight line. Experimental results on speech signals spoken by male and female speakers show that the current method can reduce the occurrence of octave errors in greater numbers when compared with other AMDF based functions.
ISCA-INST SPEECH COMMUNICATION ASSOC, English
Web of Science ID：WOS:000269965901166
Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
Rahman M.S; Tanaka H; Shimamura T
Proceedings of International Conference on Signals and Electronic Systems, 2006
Adaptive Time Variant Channel Equalization Using Phase Banded LMS Algorithm
Mohammed H.W; Shimamura T
Journal of Signal Processing, Volume：10, First page：227, Last page：230, 2006
Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
Mohammed H. W; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Active Noise Control Using Cascaded Adaptive Filters
Isozaki K; Tsuda Y; Shimamura T
Journal of Signal Processing, Volume：10, First page：279, Last page：282, 2006
Active Noise Control Using Cascaded Adaptive Filters
Isozaki K; Tsuda Y; Shimamura T
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Time delay interpolation by system response coefficient ratios J Gamba; T Shimamura
IEEE SIGNAL PROCESSING LETTERS,
Volume：12,
Number：9,
First page：641,
Last page：644, Sep. 2005
In this letter, we propose a new method of time delay interpolation based on the finite impulse response (FIR) filter coefficient ratios and derived from the Lagrange interpolation coefficients. The proposed method gives an explicit formula for the time delay that requires no frequency-domain transformations, making it suitable for entirely time-domain applications. Simulation results confirm the effectiveness of the proposed method.
IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, English
DOI：https://doi.org/10.1109/LSP.2005.853047DOI ID：10.1109/LSP.2005.853047,
ISSN：1070-9908,
eISSN：1558-2361,
Web of Science ID：WOS:000231234600012 Nonstationary noise estimation using low-frequency regions for spectral subtraction K Yamashita; T Shimamura
IEEE SIGNAL PROCESSING LETTERS,
Volume：12,
Number：6,
First page：465,
Last page：468, Jun. 2005
In this letter, a noise estimation method for spectral subtraction is proposed by using low-frequency regions of noisy speech. The method allows tracking the time variation of noise without complicated computation. The performance of a spectral subtraction method based on the new noise estimation is investigated, and its effectiveness is shown. in practical experiments.
IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, English
DOI：https://doi.org/10.1109/LSP.2005.847864DOI ID：10.1109/LSP.2005.847864,
ISSN：1070-9908,
Web of Science ID：WOS:000229157700009 Equalizer-aided time delay tracking based on L-1-normed finite differences J Gamba; T Shimamura
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES,
Volume：E88A,
Number：4,
First page：978,
Last page：987, Apr. 2005
This paper addresses the estimation of time delay between two spatially separated noisy signals by system identification modeling with the input and output corrupted by additive white Gaussian noise. The proposed method is based on a modified adaptive Butler-Cantoni equalizer that decouples noise variance estimation from channel estimation. The bias in time delay estimates that is induced by input noise is reduced by an IIR whitening filter whose coefficients are found by the Burg algorithm. For step time-variant delays, a dual mode operation scheme is adopted in which we define a normal operating (tracking) mode and an interrupt operating (optimization) mode. In the tracking mode, only a few coefficients of the impulse response vector are monitored through L-1-normed finite forward differences tracking, while in the optimization mode, the time delay optimized. Simulation results confirm the superiority of the proposed approach at low signal-to-noise ratios.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English
DOI：https://doi.org/10.1093/ietfec/e88-a.4.978DOI ID：10.1093/ietfec/e88-a.4.978,
ISSN：0916-8508,
eISSN：1745-1337,
CiNii Articles ID：10016562650,
Web of Science ID：WOS:000228412600025 Spectrum estimation by noise-compensated data extrapolation J Gamba; T Shimamura
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES,
Volume：E88A,
Number：3,
First page：702,
Last page：711, Mar. 2005
High-resolution spectrum estimation techniques have been extensively studied in recent publications. Knowledge of the noise variance is vital for spectrum estimation from noise-corrupted observations. This paper presents the use of noise compensation and data extrapolation for spectrum estimation. We assume that the observed data sequence can be represented by a set of autoregressive parameters. A recently proposed iterative algorithm is then used for noise variance estimation while autoregressive parameters are used for data extrapolation. We also present analytical results to show the exponential decay characteristics of the extrapolated samples and the frequency domain smoothing effect of data extrapolation. Some statistical results are also derived. The proposed noise-compensated data extrapolation approach is applied to both the autoregressive and FFT-based spectrum estimation methods. Finally, simulation results show the superiority of the method in terms of bias reduction and resolution improvement for sinusoids buried in noise.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English
DOI：https://doi.org/10.1093/ietfec/e88-a.3.702DOI ID：10.1093/ietfec/e88-a.3.702,
ISSN：0916-8508,
eISSN：1745-1337,
CiNii Articles ID：110003213364,
Web of Science ID：WOS:000227828700012 Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
Shimamura T; Yamauchi J
First page：47, Last page：52, 2005
Improved Spectral Subtraction Utilizing Iterative Processing
YAMASHITA K.; OGATA Shin'ya; SHIMAMURA Tetsuya
The IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences (Japanese edition) A, Volume：J88-A, Number：11, First page：1246, Last page：1257, 2005
copyright(c)2005 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html本論文では, 雑音付加音声の雑音低減の手法であるスペクトル引き算法に, 反復処理とそれに適したパラメータ設定を施した, 新しい雑音抑制技術を提案する. 反復処理とは, 一度雑音低減処理を施した推定音声を再度入力信号とみなし, 音声強調処理を施す手段であり, 残留雑音の低減が見込まれる. 反復ごとにパラメータを調整することで, 音声の劣化を抑えた更なる残留雑音低減が可能となる. また, 提案法を実行する際に, スペクトル引き算のもつリアルタイム性を保持する手法も同時に提案する. 2種類の提案法の特性を, 白色雑音, 自動車雑音, 人混み雑音を付加した実音声を用い, 従来のスペクトル引き算法及びその改良法と比較する. 主観評価及び客観評価により, 各提案法はすべての雑音環境に対して優れた結果を示すことが確認された.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003501908, CiNii Books ID：AN10013345
Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
First page：6288,
Last page：6291, 2005
DOI：https://doi.org/10.1109/ISCAS.2005.1466073DOI ID：10.1109/ISCAS.2005.1466073 Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
Gamba J; Shimamura T
First page：6, Last page：9, 2005
SIDOモデルを用いたブラインド等化に関する一検討
藤田昌宏; 津田雄亮; 島村徹也
Volume：104719, First page：37, Last page：41, 2005
音声信号のための雑音低減技術 （その２）
島村徹也
Volume：9, Number：3, First page：183, Last page：188, 2005
音声信号のための雑音低減技術 （その１）
島村徹也
Volume：9, Number：2, First page：91, Last page：98, 2005
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
島村徹也; 和田存功
Volume：6, First page：64, Last page：66, 2005
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
緒方伸哉; 島村徹也
Volume：9, Number：3, First page：255, Last page：266, 2005
Japanese
ISSN：1342-6230, CiNii Articles ID：40006817548, CiNii Books ID：AA11147833
音声強調のための反射係数を利用した雑音パワー推定
緒方伸哉; 島村徹也
Volume：9, First page：325, Last page：334, 2005
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
磯崎弘太; 津田雄介; 島村徹也
Volume：105, Number：482, First page：45, Last page：50, 2005
Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
Tsutsumi Y; Tsuda Y; Shimamura T
First page：2D-09.3, 2005
Channel Estimation Based on Classification Approaches to Equalization of Time Variant Multipath Channels
Tsutsumi Y; Tsuda Y; Shimamura T
Volume：105, Number：29, First page：47, Last page：52, 2005
Performance Improvement of a Channel Estimation Based Equalizer on Time Variant Multipath Channels
Tsuda Y; Shimamura T
First page：A4-2, 2005
Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
Tanaka H; Shimamura T
First page：404, Last page：409, 2005
Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
Shimamura T; Yamauchi J
Volume：3, First page：323, Last page：330, 2005
Variable Step-Size LMS Estimator for Fast Fading Channels
OIKAWA Shingo; TSUDA Yusuke; SHIMAMURA Tetsuya
IEICE technical report. Image engineering, Volume：105, Number：30, First page：53, Last page：58, 2005
We propose a novel channel estimation technique for fast fading channels. The proposed scheme supposes multiple linear transversal filters, which are constructed in a parallel structure. The estimator coefficients for the proposed estimation scheme are adapted by the least mean square (LMS) algorithm. To obtain a further performance improvement of the proposed scheme, a technique to adjust the step-size for the LMS is also introduced. Computer simulation results show that the proposed method provides a significant improvement related to the conventional LMS estimator in high fade rate conditions.
The Institute of Electronics, Information and Communication Engineers, English
ISSN：0913-5685, CiNii Articles ID：110003205481, CiNii Books ID：AN10013006
A Parallel Estimator with LMS Adaptation for Fast Fading Channels
Oikawa S; Tsuda Y; Shimamura T
First page：2C-12.4, 2005
Frequency Domain Magnitude Banded LMS Algorithm for Equalization of Rapidly Time Variant Channels
Mohammed H,W; Shimamura T; Cowan C.F.N
Volume：12, First page：1, Last page：6, 2005
White Noise Removal in Image by Iterative Spectral Subtraction Method
Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
First page：13, Last page：16, 2005
Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
Kawanaka R; Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
First page：5, Last page：8, 2005
Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
Shimamura T; Yamauchi J
Proceedings of WSEAS International Conference on Electronics, Control and Signal Processing, First page：47, Last page：52, 2005
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)
山下浩平; 緒方伸哉; 島村徹也
電子情報通信学会論文誌. A, 基礎・境界, Volume：J88-A, Number：11, First page：1246, Last page：1257, 2005
copyright(c)2005 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html本論文では, 雑音付加音声の雑音低減の手法であるスペクトル引き算法に, 反復処理とそれに適したパラメータ設定を施した, 新しい雑音抑制技術を提案する. 反復処理とは, 一度雑音低減処理を施した推定音声を再度入力信号とみなし, 音声強調処理を施す手段であり, 残留雑音の低減が見込まれる. 反復ごとにパラメータを調整することで, 音声の劣化を抑えた更なる残留雑音低減が可能となる. また, 提案法を実行する際に, スペクトル引き算のもつリアルタイム性を保持する手法も同時に提案する. 2種類の提案法の特性を, 白色雑音, 自動車雑音, 人混み雑音を付加した実音声を用い, 従来のスペクトル引き算法及びその改良法と比較する. 主観評価及び客観評価により, 各提案法はすべての雑音環境に対して優れた結果を示すことが確認された.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003501908, CiNii Books ID：AN10013345
Speech enhancement using a technique of adaptive bias suppression
Hirobumi Tanaka; Naomi Yamamura; Yuichiroh Ohhashi; Tetsuya Shimamura
2005 39TH ASILOMAR CONFERENCE ON SIGNALS, SYSTEMS AND COMPUTERS, VOLS 1 AND 2, First page：540, Last page：544, 2005
In this paper, we assume speech corrupted by white Gaussian noise and propose a technique of adaptive bias suppression. A linear predictor is used as the basic filter and gamma-NLMS(Normalized Least Mean Square) algorithm, which is useful for suppressing bias, is adopted for the predictor adaptation. In addition, for suppressing bias more effectively we apply a weighting parameter and filter banks to the predictor. Experiments on continuous speech result in that the proposed predictor provides superior performances.
IEEE, English
Web of Science ID：WOS:000238142000103
Voice source modeling for accurate speech analysis
M. Shahidur Rahman; Tetsuya Shimamura
2005 39TH ASILOMAR CONFERENCE ON SIGNALS, SYSTEMS AND COMPUTERS, VOLS 1 AND 2, First page：305, Last page：309, 2005
A two-pass least square method have been proposed for estimating the vocal tract parameters. An often encountered problem in using the conventional linear prediction analysis is due to the harmonic structure of the excitation source of voiced speech. This harmonic characteristic is coupled with the estimation of autoregressive (AR) coefficients that results in difficulties in estimating the vocal tract filter. This paper models the effective voice source from the residual obtained through the covariance analysis in the first-pass which is then used as input to the second-pass least square analysis. A better source-filter separation is thus achieved. The formant frequencies and bandwidths estimated using the proposed method for synthetic vowels are found to be accurate up to a factor of more than three (in percent) compared to the conventional method. Since the source characteristic is taken into account, local variations due to the positioning of analysis window are reduced significantly. The validity of the proposed method is also verified by inspecting the spectra obtained from natural vowel sounds uttered by high-pitched female speaker.
IEEE, English
Web of Science ID：WOS:000238142000057
Quality improvement of bone-conducted speech Tetsuya Shimamura; Takeshi Tomikura
Proceedings of the 2005 European Conference on Circuit Theory and Design,
Volume：3,
First page：73,
Last page：76, 2005
One method to communicate in a very high noise environment is to use bone-conducted speech. In this paper, a technique of improving the quality of bone-conducted speech is presented. The bone-conducted speech signal of a speaker is passed through a reconstruction filter designed by using the long-term spectra of the bone-conducted and normal speech signals. Properties of the normal, bone-conducted and reconstructed speech signals are investigated and it is shown that for speaker recognition, the reconstructed speech signal could be usefully utilized.
English
DOI：https://doi.org/10.1109/ECCTD.2005.1523063DOI ID：10.1109/ECCTD.2005.1523063,
SCOPUS ID：33749024465 A reconstruction filter for bone-conducted speech Tetsuya Shimamura; Toshiki Tamiya
Midwest Symposium on Circuits and Systems,
Volume：2005,
First page：1847,
Last page：1850, 2005
Bone conduction is useful as a tool to accomplish speech enhancement in noisy environments. In this paper, we design a linear phase impulse response filter to reconstruct the quality of the bone-conducted speech signal obtained from a speaker. The bone-conducted speech observation as well as the normal speech information are effectively utilized to design the filter. From experimental results, the properties of the reconstruction filter are investigated. © 2005 IEEE.
English
DOI：https://doi.org/10.1109/MWSCAS.2005.1594483DOI ID：10.1109/MWSCAS.2005.1594483,
ISSN：1548-3746,
SCOPUS ID：33847106970 Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
Proceedings of IEEE International Symposium on Circuits and Systems,
First page：6288,
Last page：6291, 2005
DOI：https://doi.org/10.1109/ISCAS.2005.1466073DOI ID：10.1109/ISCAS.2005.1466073 Linear prediction using homomorphic deconvolution in the autocorrelation domain MS Rahman; T Shimamura
2005 IEEE INTERNATIONAL SYMPOSIUM ON CIRCUITS AND SYSTEMS (ISCAS), VOLS 1-6, CONFERENCE PROCEEDINGS,
First page：2855,
Last page：2858, 2005
The conventional model of the linear prediction analysis suffers from difficulties in estimating vocal tract characteristics of high-pitched speakers. This paper shows that for voiced speech the vocal tract characteristics can be estimated accurately by homomorphic deconvolution in the autocorrelation domain. The speech autocorrelation function used by linear prediction is actually an 'aliased' version of that of the vocal tract system impulse response. This aliasing occurs due to the periodic nature of voiced speech. By using cepstrum analysis, the effect of this periodicity is eliminated from the autocorrelation function which is also periodic with the same periodicity as speech itself. The formant frequencies estimated using the deconvolved autocorrelation sequences of the system impulse response are found to be accurate by more than an order of magnitude when compared with the conventional linear prediction. The accuracy of formant estimation is verified on synthetic vowels for a wide range of pitch periods. The validity of the proposed method is also examined by inspecting the estimated spectral envelopes of real speech spoken by a female child.
IEEE, English
DOI：https://doi.org/10.1109/ISCAS.2005.1465222DOI ID：10.1109/ISCAS.2005.1465222,
ISSN：0271-4302,
Web of Science ID：WOS:000232002402237 Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
Gamba J; Shimamura T
Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing, First page：6, Last page：9, 2005
SIDOモデルを用いたブラインド等化に関する一検討
藤田昌宏; 津田雄亮; 島村徹也
電子情報通信学会技術研究報告, Volume：104719, First page：37, Last page：41, 2005
音声信号のための雑音低減技術 （その２）
島村徹也
Journal of Signal Processing, Volume：9, Number：3, First page：183, Last page：188, 2005
Japanese
ISSN：1342-6230, CiNii Articles ID：40006817541, CiNii Books ID：AA11147833
音声信号のための雑音低減技術 （その１）
島村徹也
Journal of Signal Processing, Volume：9, Number：2, First page：91, Last page：98, 2005
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
島村徹也; 和田存功
埼玉大学地域共同研究センター紀要, Volume：6, Number：6, First page：64, Last page：66, 2005
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.
Japanese
ISSN：1347-4758, CiNii Articles ID：120001371339, CiNii Books ID：AA11808968
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
緒方伸哉; 島村徹也
信号処理, Volume：9, Number：3, First page：255, Last page：266, 2005
Japanese
ISSN：1342-6230, CiNii Articles ID：40006817548, CiNii Books ID：AA11147833
音声強調のための反射係数を利用した雑音パワー推定
緒方伸哉; 島村徹也
信号処理, Volume：9, Number：4, First page：325, Last page：334, 2005
Japanese
ISSN：1342-6230, CiNii Articles ID：40006886206, CiNii Books ID：AA11147833
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
磯崎弘太; 津田雄介; 島村徹也
電子情報通信学会技術研究報告, Volume：105, Number：482, First page：45, Last page：50, 2005
Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
Tsutsumi Y; Tsuda Y; Shimamura T
Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology, First page：2D-09.3, 2005
Channel Estimation Based on Classification Approaches to Equalization of Time Variant Multipath Channels
Tsutsumi Y; Tsuda Y; Shimamura T
Technical Report of the IEICE, Volume：105, Number：29, First page：47, Last page：52, 2005
Performance Improvement of a Channel Estimation Based Equalizer on Time Variant Multipath Channels
Tsuda Y; Shimamura T
Proceedings of Signal Processing Symposium, First page：A4-2, 2005
Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
Tanaka H; Shimamura T
Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing, First page：404, Last page：409, 2005
Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
Shimamura T; Yamauchi J
WSEAS Transactions on Signal Processing, Volume：3, First page：323, Last page：330, 2005
Formant frequency estimation of high-pitched speech by homomorphic prediction M. Shahidur Rahman; Tetsuya Shimamura
Acoustical Science and Technology,
Volume：26,
Number：6,
First page：502,
Last page：510, 2005
The conventional model of the linear prediction analysis suffers from difficulties in estimating vocal tract characteristics of high-pitched speakers. This is because the autocorrelation function used by the autocorrelation method of linear prediction for estimating autoregressive coefficients is actually an "aliased" version of that of the vocal tract impulse response. This "aliasing" occurs due to the periodic nature of voiced speech. Generally it is accepted that homomorphic filtering can be used to obtain an estimate of vocal tract impulse response which is free from periodicity. Thus linear prediction of the resulting vocal tract impulse response (referred to as homomorphic prediction) is expected to be free from variations of fundamental frequencies. To our knowledge any experimental study, however, has not yet appeared on the suitability of this method for analyzing high-pitched speech. This paper presents a detail study on the prospects of homomorphic prediction as a formant tracking tool especially for high-pitched speech where linear prediction fails to obtain accurate estimation. The formant frequencies estimated using the proposed method are found to be accurate by more than an order of magnitude compared to the conventional procedure. The accuracy of formant estimation is verified on synthetic vowels for a wide range of pitch periods covering typical male and high-pitched female speakers. The validity of the proposed method is also examined by inspecting the spectral envelopes of natural speech spoken by high-pitched female speakers. We noticed that almost all the previous methods dealing with this limitation of linear prediction are based on the covariance technique where the obtained AR filter can be unstable. The solutions obtained by the current method are guaranteed to be stable which makes it superior for many speech analysis applications.
English
DOI：https://doi.org/10.1250/ast.26.502DOI ID：10.1250/ast.26.502,
ISSN：1346-3969,
CiNii Articles ID：110003143601,
SCOPUS ID：27644474210 Variable Step-Size LMS Estimator for Fast Fading Channels
Oikawa S; Tsuda Y; Shimamura T
Technical Report of the IEICE, Volume：105, Number：30, First page：53, Last page：58, 2005
We propose a novel channel estimation technique for fast fading channels. The proposed scheme supposes multiple linear transversal filters, which are constructed in a parallel structure. The estimator coefficients for the proposed estimation scheme are adapted by the least mean square (LMS) algorithm. To obtain a further performance improvement of the proposed scheme, a technique to adjust the step-size for the LMS is also introduced. Computer simulation results show that the proposed method provides a significant improvement related to the conventional LMS estimator in high fade rate conditions.
The Institute of Electronics, Information and Communication Engineers, English
ISSN：0913-5685, CiNii Articles ID：110003205481, CiNii Books ID：AN10013006
A Parallel Estimator with LMS Adaptation for Fast Fading Channels
Oikawa S; Tsuda Y; Shimamura T
Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology, First page：2C-12.4, 2005
Frequency Domain Magnitude Banded LMS Algorithm for Equalization of Rapidly Time Variant Channels
Mohammed H,W; Shimamura T; Cowan C.F.N
WSEAS Transactions on Electronics, Volume：12, First page：1, Last page：6, 2005
White Noise Removal in Image by Iterative Spectral Subtraction Method
Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：13, Last page：16, 2005
Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
Kawanaka R; Kobayashi T; Shimamura T; Hosoya T; Takahashi Y
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, First page：5, Last page：8, 2005
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
田中啓文; 早川晴子; 島村徹也
Volume：J87-A, Number：7, First page：899, Last page：912, 2004
ピッチ同期加算処理を用いた雑音低減に基づくLPC分析
島村徹也; 黒岩世進伸
Volume：J87, Number：A4, First page：458, Last page：469, 2004
A New Method of Noise Variance Estimation from Low-Order Yule-Walker Equations (Digital Signal Processing)
島村徹也
Volume：E87-A, Number：1, First page：270, Last page：274, 2004
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-
津田雄亮; 島村徹也
First page：B4-3, 2004
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
小林徹也; 島村徹也; 細谷徹夫; 高橋由武
2004
Nonlinear Predictive Analysis of Speech by Iterative Approach
Tanaka H; Shimamura T
First page：2055, Last page：2058, 2004
Reconstruction Filter design for Bone-Conducted Speech
Tamiya T; Shimamura T
First page：1, Last page：4, 2004
Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
Shimamura T; Yamauchi J
First page：2305, Last page：2309, 2004
Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
Shimamura T; Takagi H
First page：II1 41-144, 2004
Pitch synchronous addition and extension for linear predictive analysis of noisy speech
T Shimamura
NORSIG 2004: PROCEEDINGS OF THE 6TH NORDIC SIGNAL PROCESSING SYMPOSIUM, Volume：46, First page：196, Last page：199, 2004
This paper proposes an approach for pitch synchronous linear predictive coding (LPC) of speech in noisy environments. A not. se reduction method is derived which produces an enhanced speech signal with one pitch period. For the proposed LPC method, the enhanced one pitch speech signal is used in a form of pitch extension so that the autocorrelation function is obtained accurately. Simulation experiments show that the proposed LPC method provides a superior performance in white noise.
HELSINKI UNIVERSITY TECHNOLOGY, English
ISSN：1458-6401, Web of Science ID：WOS:000225463400050
Adaptive Non-linear Prediction of Speech in Impulse Noise
Ohhashi Y; Tanaka H; Shimamura T
First page：1675, Last page：1678, 2004
Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
Ogata S; Ebata S; Shimamura T
First page：3037, Last page：3040, 2004
A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
Mohammed H.W; Shimamura T
First page：433, Last page：438, 2004
Equalizer-aided time delay tracking based on finite differences
J. Gamba; Y. Tsuda; T. Shimamura
First page：B4-4, 2004
An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
Gamba J; Tsuda Y; Shimamura T
First page：265, Last page：268, 2004
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
田中啓文; 早川晴子; 島村徹也
電子情報通信学会論文誌. A, 基礎・境界, Volume：J87-A, Number：7, First page：899, Last page：912, 2004
ピッチ同期加算処理を用いた雑音低減に基づくLPC分析
島村徹也; 黒岩世進伸
電子情報通信学会論文誌, Volume：J87, Number：A4, First page：458, Last page：469, 2004
A New Method of Noise Variance Estimation from Low-Order Yule-Walker Equations (Digital Signal Processing)
島村徹也
IEICE transactions on fundamentals of electronics, communications and computer sciences, Volume：E87-A, Number：1, First page：270, Last page：274, 2004
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-
津田雄亮; 島村徹也
Proc. 第19回信号処理シンポジウム講演論文集, First page：B4-3, 2004
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
小林徹也; 島村徹也; 細谷徹夫; 高橋由武
電子情報通信学会技術研究報告, 2004
Nonlinear Predictive Analysis of Speech by Iterative Approach
Tanaka H; Shimamura T
Proc. 12th Europian Signal Processing Conf., First page：2055, Last page：2058, 2004
Reconstruction Filter design for Bone-Conducted Speech
Tamiya T; Shimamura T
Proceedings of International Conference on Spoken Language Processing, First page：1, Last page：4, 2004
Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
Shimamura T; Yamauchi J
Proceedings of Asilomar Conference on Signals, Systems and Computers, First page：2305, Last page：2309, 2004
Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
Shimamura T; Takagi H
Proc. 47th IEEE International Midwest Symposium on Circuits and Systems, First page：II1 41-144, 2004
Pitch Synchronous Addition and Extension for Linear Predictive Analysis of Noisy Speech
Shimamura T; Kuroiwa Y
Proc. 6th Nordic Signal Processing Symposium, First page：196, Last page：199, 2004
Adaptive Non-linear Prediction of Speech in Impulse Noise
Ohhashi Y; Tanaka H; Shimamura T
Proc. 18th International Congress on Acoustics, First page：1675, Last page：1678, 2004
Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
Ogata S; Ebata S; Shimamura T
Proc. 18th International Congress on Acoustics, First page：3037, Last page：3040, 2004
A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
Mohammed H.W; Shimamura T
Proceedings of IFAC Workshop on Adaptation and Learning in Control and Signal Processing, First page：433, Last page：438, 2004
Equalizer-aided time delay tracking based on finite differences
J. Gamba; Y. Tsuda; T. Shimamura
Proc. 19th IEICE Signal Processing Symposium, First page：B4-4, 2004
An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
Gamba J; Tsuda Y; Shimamura T
Proceedings of IEEE International Symposium on Circuits and Systems, First page：265, Last page：268, 2004
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)
島村徹也; 高木浩司
Volume：J86-A, Number：11, First page：1097, Last page：1107, 2003
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)
島村徹也; 高木浩司
電子情報通信学会論文誌. A, 基礎・境界, Volume：J86-A, Number：11, First page：1097, Last page：1107, 2003
Noise estimation using high frequency regions for spectral subtraction
J Yamauchi; T Shimamura
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, Volume：E85A, Number：3, First page：723, Last page：727, Mar. 2002
This paper presents an improved spectral subtraction method for speech enhancement. A new noise estimation method is derived in which the noise is assumed to be white. By using the property that a white noise spectrum is flat, high frequency components of a noisy speech spectrum arc, averaged and the standard deviation of the noise is estimated. This operation is performed in the analysis segment, thus the spectral subtraction method combined with the new noise estimation method does not need non-speech segments and as a result can adapt to non-stationary noise conditions. The effectiveness of the proposed spectral subtraction method is confirmed by experiments.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English, Report scientific journal
ISSN：0916-8508, eISSN：1745-1337, Web of Science ID：WOS:000174258000021
Amplitude banded RLS approach to time variant channel equalization
T Shimamura; CFN Cowan
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, Volume：E84A, Number：11, First page：2946, Last page：2949, Nov. 2001
This paper proposes a non-linear adaptive algorithm, the amplitude banded RLS (ABRLS) algorithm, as an adaptation procedure for time variant channel equalizers. In the ABRLS algorithm, a coefficient matrix is updated based on the amplitude level of the received sequence. To enhance the capability of tracking for the ABRLS algorithm, a parallel adaptation scheme is utilized which involves the structures of decision feedback equalizer (DFE). Computer simulations demonstrate that the novel ABRLS based equalizer provides a significant improvement relative to the conventional RLS DFE on a rapidly time variant communication channel.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English, Report scientific journal
ISSN：0916-8508, eISSN：1745-1337, Web of Science ID：WOS:000172132200044
Weighted autocorrelation for pitch extraction of noisy speech T Shimamura; H Kobayashi
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING,
Volume：9,
Number：7,
First page：727,
Last page：730, Oct. 2001
In this paper, we propose a modified version of the autocorrelation pitch extraction method well known to be robust against noise. Utilizing that the average magnitude difference function (AMDF) has similar characteristics with the autocorrelation function, the autocorrelation function is weighted by the reciprocal of the AMDF By simulation experiments, it is shown that the proposed pitch extraction method is useful in noisy environments.
IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, English
DOI：https://doi.org/10.1109/89.952490DOI ID：10.1109/89.952490,
ISSN：1063-6676,
Web of Science ID：WOS:000171193600002 A fast converging RLS equaliser
T Shimamura
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, Volume：E84A, Number：2, First page：675, Last page：680, Feb. 2001
It is well known that based on the structure of a transversal filter, the RLS equaliser provides the fastest convergence in stationary environments. This paper addresses an adaptive transversal equaliser which has the potential to provide more faster convergence than the RLS equaliser. A comparison is made with respect to computational complexity required for each update of equaliser coefficients, and computer simulations are demonstrated to show the superiority of the proposed equaliser.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English, Report scientific journal
ISSN：1745-1337, Web of Science ID：WOS:000166826000039
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
島村徹也; 趙奇方; 高橋淳一; 鈴木誠史
Volume：J84-A, Number：6, First page：745, Last page：758, 2001
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)
島村徹也; 吉尾重治; 趙奇方; 鈴木誠史
Volume：J84-A, Number：3, First page：436, Last page：440, 2001
Equalisation of Time Variant Multipath Channels Using Amplitude Banded LMS Algorithms(Digital Signal Processing)(Regular Section)
島村徹也
Volume：E84-A, Number：3, First page：802, Last page：812, 2001
IIR-Type Adaptive Equalizer with AR Prefilter and IIR-Type Wiener Filter
SHIMAMURA Tetsuya; SUZUKI Jouji
The Transactions of the Institute of Electronics,Information and Communication Engineers. A, Volume：J84-A, Number：1, First page：109, Last page：112, 2001
copyright(c)2001 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html筆者らは, 先にARプレフィルタを用いたIIR型の適応等化器を提案している.本論文では, そのIIR型適応等化器の特性を解析し, IIR型ウィーナーフィルタとの関係を明らかにする.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003313621, CiNii Books ID：AN10013345
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
島村徹也; 趙奇方; 高橋淳一; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J84-A, Number：6, First page：745, Last page：758, 2001
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)
島村徹也; 吉尾重治; 趙奇方; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J84-A, Number：3, First page：436, Last page：440, 2001
Equalisation of Time Variant Multipath Channels Using Amplitude Banded LMS Algorithms(Digital Signal Processing)(Regular Section)
島村徹也
IEICE transactions on fundamentals of electronics, communications and computer sciences, Volume：E84-A, Number：3, First page：802, Last page：812, 2001
A Fast Converging RLS Equaliser
島村徹也
IEICE transactions on fundamentals of electronics, communications and computer sciences, Volume：E84-A, Number：2, First page：675, Last page：680, 2001
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
島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J84-A, Number：1, First page：109, Last page：112, 2001
copyright(c)2001 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html筆者らは, 先にARプレフィルタを用いたIIR型の適応等化器を提案している.本論文では, そのIIR型適応等化器の特性を解析し, IIR型ウィーナーフィルタとの関係を明らかにする.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003313621, CiNii Books ID：AN10013345
An ARMA prefiltering approach to adaptive equalization
T Shimamura; T Takada; J Suzuki
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, Volume：E83A, Number：10, First page：2035, Last page：2039, Oct. 2000
In this paper, we propose an adaptive IIR equalizer based on prefiltering techniques. The proposed equalizer has a cascade structure of an ARMA prefilter and an adaptive FIR equalizer. The ARMA prefilter is designed based on the transfer function estimated by the gradient-type instrumental variable algorithm. Simulation results are shown to confirm the performance of the proposed adaptive IIR equalizer. key words: prefilter, instrumental variable algorithm, adaptive IIR equalizer.
IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, English, Report scientific journal
ISSN：0916-8508, eISSN：1745-1337, Web of Science ID：WOS:000090137600027
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)
有馬由紀; 島村徹也
Volume：J83-A, Number：12, First page：1455, Last page：1466, 2000
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)
有馬由紀; 島村徹也
電子情報通信学会論文誌. A, 基礎・境界, Volume：J83-A, Number：12, First page：1455, Last page：1466, 2000
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
小林載; 島村徹也
Volume：J82-A, Number：7, First page：1115, Last page：1122, 1999
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
小林載; 島村徹也
電子情報通信学会論文誌. A, 基礎・境界, Volume：J82-A, Number：7, First page：1115, Last page：1122, 1999
copyright(c)1999 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html音声の基本周波数は, 音声処理の幅広い分野で必要とされる特徴パラメータである. 音声信号から基本周波数を抽出する手法は過去に多数提案されているが, あらゆる条件に対し有効な手法はいまだに確立されていない. 本論文では, ケプストラム法を改良することにより雑音環境下における音声に有効となる基本周波数抽出法を提案する. 本手法の特色は, 対数スペクトルのうち特に雑音の影響を受けやすい高周波数成分とスペクトルの谷の部分を除去し, 音声信号の調波構造を明確にした上でケプスドラムを求める点にある. 計算機シミュレーション実,験の結果, 従来法に比べ, 本手法における抽出精度はgross pitch errorを改善することができた. 特に, 周期性を有する雑音が混入された音声の場合に, 本手法により顕著な効果が得られた.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003313374, CiNii Books ID：AN10013345
Improvement of LPC Analysis of Speech by Noise Compensation
ZHAO Qifang; SHIMAMURA Tetsuya; SUZUKI Jouji
The Transactions of the Institute of Electronics,Information and Communication Engineers. A, Volume：J81-A, Number：11, First page：1583, Last page：1591, 1998
copyright(c)1998 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html白色雑音の混入した音声信号から自己相関法を用いて予測係数を求める場合, 自己相関関数R(k)のk=0の付近にその雑音の影響は集中する.従って, 原理的には, 自己相関関数R(0)から雑音パワーを引くことにより雑音補正が行われ, LPC分析の耐雑音性が向上される.しかし, 従来の雑音補正法ではパワーの引きすぎが原因でLPCフィルタが不安定になることがある.従って, その実際的応用は困難と思われる.一方で, スペクトル推定における同様な問題を解決するため, 島村らは反復アルゴリズムを利用した改良雑音補正AR係数推定法を提案した.本論文ではこの改良雑音補正AR係数推定法を音声のLPC分析の雑音低減に応用する.評価実験からこの方法はプリエンファシス(pre-emphasis)されていない音声に対しては有効であるが, プリエンファシスされた音声に対して改善が見られないことが明らかとなった.その理由として, プリエンファシスによって雑音の影響が自己相関関数R(k)のk=1の部分にも及んだことを理論解析で示した。そして, プリエンファシスに影響されない, R(0)とR(1)の双方から雑音パワーを引き去る反復アルゴリズムを導出した.その有効性を計算機シミュレーションで確認している.更にこの方法を拡張し, 白色雑音以外の雑音にも対応できる改善法を提案している.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003312764, CiNii Books ID：AN10013345
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
金子信一郎; 鈴木誠史; 島村徹也
Volume：J81-D2, Number：7, First page：1501, Last page：1509, 1998
高速スタートアップ等化のためのButler-Cantoni法の適応化
島村徹也; 鈴木誠史
Volume：J81-A, Number：4, First page：622, Last page：630, 1998
雑音補正による音声のLPC分析の改善
島村徹也; 趙奇方; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J81-A, Number：11, First page：1583, Last page：1591, 1998
copyright(c)1998 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html白色雑音の混入した音声信号から自己相関法を用いて予測係数を求める場合, 自己相関関数R(k)のk=0の付近にその雑音の影響は集中する.従って, 原理的には, 自己相関関数R(0)から雑音パワーを引くことにより雑音補正が行われ, LPC分析の耐雑音性が向上される.しかし, 従来の雑音補正法ではパワーの引きすぎが原因でLPCフィルタが不安定になることがある.従って, その実際的応用は困難と思われる.一方で, スペクトル推定における同様な問題を解決するため, 島村らは反復アルゴリズムを利用した改良雑音補正AR係数推定法を提案した.本論文ではこの改良雑音補正AR係数推定法を音声のLPC分析の雑音低減に応用する.評価実験からこの方法はプリエンファシス(pre-emphasis)されていない音声に対しては有効であるが, プリエンファシスされた音声に対して改善が見られないことが明らかとなった.その理由として, プリエンファシスによって雑音の影響が自己相関関数R(k)のk=1の部分にも及んだことを理論解析で示した。そして, プリエンファシスに影響されない, R(0)とR(1)の双方から雑音パワーを引き去る反復アルゴリズムを導出した.その有効性を計算機シミュレーションで確認している.更にこの方法を拡張し, 白色雑音以外の雑音にも対応できる改善法を提案している.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003312764, CiNii Books ID：AN10013345
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
金子信一郎; 鈴木誠史; 島村徹也
電子情報通信学会論文誌. D-II, 情報・システム, II-情報処理, Volume：J81-D2, Number：7, First page：1501, Last page：1509, 1998
高速スタートアップ等化のためのButler-Cantoni法の適応化
島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J81-A, Number：4, First page：622, Last page：630, 1998
CiNii Articles ID：10015530445
Equalisation of time-variant communications channels via channel estimation based approaches T Shimamura; S Semnani; CFN Cowan
SIGNAL PROCESSING,
Volume：60,
Number：2,
First page：181,
Last page：193, Jul. 1997
For the purpose of tackling the problem of equalisation in a time-variant environment, two novel approaches are developed in which the separation of the channel estimation process from the equalisation process is attempted. Two linear filters - channel estimator and equalisation filter - are used to reconstruct the transmitted sequence. The gradient algorithm with degree-1 least square fading memory prediction is adopted for the channel estimator. Using the results produced by the channel estimator, the coefficients of the equalisation filter are indirectly updated. Computer simulation results show that the two channel estimation based adaptive equalisers provide significant improvement in the case of a second order Markov communication channel model. (C) 1997 Elsevier Science B.V.
ELSEVIER SCIENCE BV, English
DOI：https://doi.org/10.1016/S0165-1684(97)00071-6DOI ID：10.1016/S0165-1684(97)00071-6,
ISSN：0165-1684,
CiNii Articles ID：80009850376,
Web of Science ID：WOS:A1997XU66900004 品質劣化音声のためのLPC分析の一改良法
島村徹也; 國枝伸行; 鈴木誠史
Volume：J80-A, Number：9, First page：1564, Last page：1566, 1997
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
國枝伸行; 島村徹也; 鈴木誠史
Volume：J80-A, Number：3, First page：435, Last page：443, 1997
品質劣化音声のためのLPC分析の一改良法
島村徹也; 國枝伸行; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J80-A, Number：9, First page：1564, Last page：1566, 1997
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
國枝伸行; 島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J80-A, Number：3, First page：435, Last page：443, 1997
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
國枝伸行; 島村徹也; 鈴木誠史
Volume：J79-A, Number：3, First page：541, Last page：550, 1996
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
國枝伸行; 島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J79-A, Number：3, First page：541, Last page：550, 1996
copyright(c)1996 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html我々は相関関数を利用した雑音低減方式について検討している. 先に提案したSPACやSPADは2乗ひずみや高調波ひずみを生じ, 明瞭性を低下させる欠点があった. また前向き後向き差分関数は, ランダム雑音に埋もれた単一正弦波をこうしたひずみを生じることなく強調できる. しかしながら, この関数を音声のような複合波に適用するとひずみが生じる欠点があった. 本論文では, 前向き後向き差分関数を音声強調に利用するため, フィルタバンクを利用した手法を提案する. 本方式では, 音声が調波構造からなることに着目する. すなわち, 音声をフィルタバンクを利用して複数の帯域制限信号に分解し, それぞれの信号に対して前向き後向き差分関数によって高調波を強調する. 強調した高調波を再合成することによって, 音声強調が実現できる. 本論文では, まず前向き後向き差分関数が帯域制限信号に対しても有効であることを示す. そして, 帯域幅が100 Hzのフィルタバンクを構成した提案法により, 疑似音声のSN比を6〜7dB改善できることを計算機シミュレーシヨンによって求めた. 本方式の実際の音声に対する効果を調べた結果, スペクトル包絡のピークを強調できることを確認できた. 試聴実験の結果からは, 提案法がSRACやSPADで生じるひずみを抑え, 雑音低減できるという結果が得られた.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003312403, CiNii Books ID：AN10013345
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
島村徹也; 繆衛国; 鈴木誠史
Volume：J78-A, Number：8, First page：965, Last page：976, 1995
ブラインド等化のためのプレフィルタリング
伊藤克子; 島村徹也; 鈴木誠史
Volume：J78-A, Number：3, First page：323, Last page：331, 1995
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
島村徹也; 繆衛国; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J78-A, Number：8, First page：965, Last page：976, 1995
ブラインド等化のためのプレフィルタリング
伊藤克子; 島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J78-A, Number：3, First page：323, Last page：331, 1995
Enhancement of Single Sinusoidal Signal by Forward and Backward Difference Function
KUNIEDA Nobuyuki; SHIMAMURA Tetsuya; SUZUKI Jouji
The Transactions of the Institute of Electronics,Information and Communication Engineers. A, Volume：J77-A, Number：9, First page：1307, Last page：1311, 1994
copyright(c)1994 IEICE許諾番号：07RB0174雑音中の単一正弦波をひずみなく強調するための新しい関数として前向き後向き差分関数を定義する.白色雑音の重畳した正弦波を対象に,この関数によるSN比改善特性を求めたところ,自己相関関数よりも優れた効果を得ることができた.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003313184, CiNii Books ID：AN10013345
Burg 法のためのデータ予測
島村徹也; 鈴木誠史
Volume：J77-A, Number：8, First page：1182, Last page：1185, 1994
前向き後向き差分関数による単一正弦波信号の強調
國枝伸行; 島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J77-A, Number：9, First page：1307, Last page：1311, 1994
copyright(c)1994 IEICE許諾番号：07RB0174雑音中の単一正弦波をひずみなく強調するための新しい関数として前向き後向き差分関数を定義する.白色雑音の重畳した正弦波を対象に,この関数によるSN比改善特性を求めたところ,自己相関関数よりも優れた効果を得ることができた.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003313184, CiNii Books ID：AN10013345
Burg 法のためのデータ予測
島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J77-A, Number：8, First page：1182, Last page：1185, 1994
全極型プレフィルタを用いた IIR 型適応等化器
伊藤克子; 島村徹也; 八嶋弘幸; 鈴木誠史
Volume：J76-A, Number：9, First page：1279, Last page：1285, 1993
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)
岩間智史; 島村徹也; 鈴木誠史
Volume：J76-A, Number：6, First page：910, Last page：912, 1993
全極型プレフィルタを用いた IIR 型適応等化器
伊藤克子; 島村徹也; 八嶋弘幸; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J76-A, Number：9, First page：1279, Last page：1285, 1993
copyright(c)1993 IEICE許諾番号：07RB0174本論文では,全極型プレフィルタを用いたIIR型適応等化器を提案する.本等化器は,通信路のひずみが大きく,悪条件により等化が困難な場合においても,LMSアルゴリズムを用いて,その等化を可能にする.等化器の構成は,FIR型適応フィルタの前部に推定された通信路の逆フィルタを置く縦続2段構成であり,全体の構成はIIRシステムである.FIR型適応フィルタのみを用いた従来のFIR型適応等化器では,入力信号の相関が大きい場合のLMSアルゴリズムの収束特性の劣化が問題となっていた.本法では,通信路で生じた信号の相関を低減する効果を有する逆フィルタをFIR型適応フィルタの前部に置くことにより,LMSアルゴリズムの収束特性を改善し,トレーニング時間の短縮を実現している.また,構成をIIR型にすることにより,FIR型の場合より低次のシステムで精度の高い等化が可能になる.システムの安定性は,簡単な操作により保証される.計算機シミュレーションでは,提案する等化器の有効性を立証する.
The Institute of Electronics, Information and Communication Engineers, Japanese
ISSN：0913-5707, CiNii Articles ID：110003312945, CiNii Books ID：AN10013345
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)
岩間智史; 島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J76-A, Number：6, First page：910, Last page：912, 1993
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
島村徹也; 鈴木誠史
Volume：J75-A, Number：12, First page：1783, Last page：1791, 1992
Speech Processing System by Use of Auto-Difference Function - SPAD -
KUNIEDA Nobuyuki; SHIMAMURA Tetsuya; SUZUKI Jouji; YASHIMA Hiroyuki
The Transactions of the Institute of Electronics,Information and Communication Engineers. A, Volume：J75-A, Number：11, First page：1769, Last page：1772, 1992
copyright(c)1992 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html雑音の重畳した音声の短時間差分関数の波形を逐次接続して出力することにより，雑音レベルの低減を行うことができる．この方式は，従来の自己相関関数を利用した音声処理方式（SPAC）よりも単純な構成で，同程度の雑音低減ができることを示す．
Japanese
ISSN：0913-5707, CiNii Articles ID：10006928407, CiNii Books ID：AN10013345
Least Squares Adaptive Filter with Complex Coefficients for Ill-Conditioned Adaptive Equalization
SHIMAMURA Tetsuya; SUZUKI Jouji
The Transactions of the Institute of Electronics,Information and Communication Engineers. A, Volume：J75-A, Number：11, First page：1666, Last page：1674, 1992
copyright(c)1992 IEICE許諾番号：07RB0174本論文では，チャネルのひずみが大きい悪条件下においても，良好な等化を実現することが可能な最小2乗型適応等化器を提案している．本法は，トランスバーサル型等化器のタップ係数を複素係数化し，次数を半減することにより，等化器への入力信号からなる入力相関行列の条件数が低減できることに着目したものである．具体的には，トレーニングモードにおいて，等化器の入出力信号を負の周波数成分を含まない解析信号に変換し，更に比2のデシメーション操作を施した後，複素係数適応フィルタリングを実行する．従来の実係数フィルタリング法に比べ，推定されるタップ係数値の精度が大幅に向上し，また，単位時間当りに必要とされる演算量が低減される．理論解析および計算機シミュレーション実験を通して，提案法の有効性が立証される．
Japanese
ISSN：0913-5707, CiNii Articles ID：10006928256, CiNii Books ID：AN10013345
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
島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J75-A, Number：12, First page：1783, Last page：1791, 1992
差分関数を利用した音声処理法式 - SPAD -
國枝伸行; 島村徹也; 鈴木誠史; 八嶋弘幸
電子情報通信学会論文誌. A, 基礎・境界, Volume：J75-A, Number：11, First page：1769, Last page：1772, 1992
copyright(c)1992 IEICE許諾番号：07RB0174 http://www.ieice.org/jpn/trans_online/index.html雑音の重畳した音声の短時間差分関数の波形を逐次接続して出力することにより，雑音レベルの低減を行うことができる．この方式は，従来の自己相関関数を利用した音声処理方式（SPAC）よりも単純な構成で，同程度の雑音低減ができることを示す．
Japanese
ISSN：0913-5707, CiNii Articles ID：10006928407, CiNii Books ID：AN10013345
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
島村徹也; 鈴木誠史
電子情報通信学会論文誌. A, 基礎・境界, Volume：J75-A, Number：11, First page：1666, Last page：1674, 1992
copyright(c)1992 IEICE許諾番号：07RB0174本論文では，チャネルのひずみが大きい悪条件下においても，良好な等化を実現することが可能な最小2乗型適応等化器を提案している．本法は，トランスバーサル型等化器のタップ係数を複素係数化し，次数を半減することにより，等化器への入力信号からなる入力相関行列の条件数が低減できることに着目したものである．具体的には，トレーニングモードにおいて，等化器の入出力信号を負の周波数成分を含まない解析信号に変換し，更に比2のデシメーション操作を施した後，複素係数適応フィルタリングを実行する．従来の実係数フィルタリング法に比べ，推定されるタップ係数値の精度が大幅に向上し，また，単位時間当りに必要とされる演算量が低減される．理論解析および計算機シミュレーション実験を通して，提案法の有効性が立証される．
Japanese
ISSN：0913-5707, CiNii Articles ID：10006928256, CiNii Books ID：AN10013345
Equalization of Time-Variant Communications Channels via Adaptive AR Prefiltering
2009
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
2009
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
2009
反復適応ウィナーフィルタを用いた画像復元
2009
Dual Adaptive Pre-Whitening Filters for LMS Algorithm
2009
Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
2009
Image Restoration via Wiener Filtering with Improved Noise Estimation
2009
Equalization of Time-Variant Communications Channels via Adaptive AR Prefiltering
Proceedings of WRI World Congress on Computer Science and Information Engineering, 2009
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2009総合大会講演論文集, 2009
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2009総合大会講演論文集, 2009
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2009総合大会講演論文集, 2009
Dual Adaptive Pre-Whitening Filters for LMS Algorithm
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2009
Time Delay Estimation of Arrival and Spectrum Subtraction for Acoustic Dual-Microphone Systems
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2009
Image Restoration via Wiener Filtering with Improved Noise Estimation
Proceedings of WSEAS International Conference on Signal Processing, Robotics and Automation, 2009
Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization
2008
High Pitch Source Isolation Using Complex Cepstrum in the Autocorrelation Domain
2008
双対の適応白色化フィルタを用いたLMSアルゴリズム
2008
線形予測誤差を用いた骨導音声の品質改善
2008
Poster presentation
An Efficient and Effective Variable Step Size NLMS Algorithm
2008
Adaptive Line Enhancer for Time Delay Estimation of Ultrasonic Echoes
2008
Amplitude-Division Parallel LMS Estimator
2008
Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm
2008
高騒音環境下における骨導音声を用いた話者認識
2008
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
2008
Poster presentation
縦続型適応非線形予測器を用いた音声信号の予測分析
2008
Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
2008
Iterative Cross-Correlation Method for Time Delay Estimation
2008
Bone-Conducted Speech for Speaker Verification
2008
Wavelet Based Denoising for Images Degraded by Poisson Noise
2008
Amplitude Banded Technique and Parallel Structure for Godard and Sato Algorithms of Blind Channel Equalization
Proceedings of International Conference on Computer and Information Technology, 2008
High Pitch Source Isolation Using Complex Cepstrum in the Autocorrelation Domain
Proceedings of IEEE Asia Pacific Conference on Circuits and Systems, 2008
双対の適応白色化フィルタを用いたLMSアルゴリズム
信号処理シンポジウム講演論文集, 2008
線形予測誤差を用いた骨導音声の品質改善
信号処理シンポジウム講演論文集, 2008
Poster presentation
An Efficient and Effective Variable Step Size NLMS Algorithm
Proceedings of Asilomar Conference on Signals, Systems and Computers, 2008
Adaptive Line Enhancer for Time Delay Estimation of Ultrasonic Echoes
Proceedings of International Symposium on Communication and Information Technology, 2008
Amplitude-Division Parallel LMS Estimator
Proceedings of IEEE International Midwest Symposium on Circuits and Systems, 2008
Convergence Evaluation of a Variable Step-Size LMSE Adaptive Switching Algorithm
Proceedings of IEEE International Networking and Communications Conference, 2008
高騒音環境下における骨導音声を用いた話者認識
日本音響学会春期研究発表会講演論文集, 2008
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
日本音響学会春期研究発表会講演論文集, 2008
Poster presentation
縦続型適応非線形予測器を用いた音声信号の予測分析
日本音響学会春期研究発表会講演論文集, 2008
Quality Improvement of Bone-Conducted Speech Using Linear Prediction Analysis Filter
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2008
Iterative Cross-Correlation Method for Time Delay Estimation
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2008
Bone-Conducted Speech for Speaker Verification
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2008
Wavelet Based Denoising for Images Degraded by Poisson Noise
Proceedings of IASTED International Conference on Biomedical Engineering, 2008
Amplitude Banded Sato Algorithm for Blind Channel Equalization
2007
Poster presentation
Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
2007
Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
2007
Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
2007
Poster presentation
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
2007
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
2007
Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
2007
Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
2007
Indirect Cross-Correlation Method for Time Delay Estimation
2007
Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
2007
Discrete Cosine Transform Domain Parallel LMS Equalizer
2007
Amplitude Banded Sato Algorithm for Blind Channel Equalization
Proceedings of IEEE International Conference on Signal Processing and Communications, 2007
Poster presentation
Image Denoising for Poisson Noise by Pixel Values Based Division and Wavelet Shrinkage
Proceedings of International Symposium on Nonlinear Theory and Its Applications, 2007
Equalization with Amplitude Banded LMS Adaptation for Stationary Channels
Proceedings of IASTED International Conference on Signal and Image Processing, 2007
Performance of the Amplitude Banded LMS Equalizer on Stationary Channels
Proceedings of IEEE International Workshop on Nonlinear Dynamics of Electronic Systems, 2007
Poster presentation
Discrete Wavelet Based Keyframe Extraction Method from Motion Capture Data
Proceedings of NICOGRAPH International 2007, 2007
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
電子情報通信学会総合大会講演論文集, 2007
Restoration of Bone-Conducted Speech with the Wiener Filter and Long-Term Spectra
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2007
Magnitude Spectral Subtraction with DFTLMS Algorithm for Adaptive Line Enhancer
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2007
Indirect Cross-Correlation Method for Time Delay Estimation
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2007
Adaptive Non-Linear Prediction with Order Statistics for Speech in Impulsive Noise
Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications, 2007
Discrete Cosine Transform Domain Parallel LMS Equalizer
Proceedings on WSEAS International Conference on Circuits, Systems, Signal and Telecommunications, 2007
A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels
2006
Adaptive Non-Linear Prediction with Order Statistics for Speech Signals in Mixture Noise
2006
Active Noise Control Using A Refined Filtering Approach
2006
Wavelet Based Keyframe Extraction Method from Motion Capture Data
2006
Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
2006
Learning for Bone-Conducted Speech via Adaptive and Neural Filters
2006
Pitch Determination using Aligned AMDF
2006
A Harsh Noise Assessment Measure for Speech Enhancement
2006
Improving Bone-Conducted Speech Quality via Neural Network
2006
SIDOモデルを用いたブラインド等化に関する一検討
2006
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
2006
Reconstruction Filter design for Bone-Conducted Speech
2006
骨導音声の品質改善について （その１）
2006
Poster presentation
反復RFアルゴリズムを利用したリアルタイム騒音制御
2006
Poster presentation
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
2006
線形予測フィルタを用いた適応音声強調
2006
Poster presentation
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
2006
Poster presentation
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
2006
Poster presentation
A Study on Normalized LMS Algorithm Using Refined Filtering Technique
2006
Noise Estimation Using Multifrequency Regions for Spectral Subtraction
2006
Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
2006
Active Noise Control Using Cascaded Adaptive Filters
2006
A Parallel Estimator with LMS and RLS Adaptation for Fast Fading Channels
Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems, 2006
Adaptive Non-Linear Prediction with Order Statistics for Speech Signals in Mixture Noise
Proceedings of IEEE International Symposium on Intelligent Signal Processing and Communication Systems, 2006
Active Noise Control Using A Refined Filtering Approach
Proceedings of the 35th International Congress and Exposition on Noise Control Engineering, 2006
Wavelet Based Keyframe Extraction Method from Motion Capture Data
Proceedings on Asia Digital Art and Design Association, 2006
Autoregressive Moving Average (ARMA) Analysis of Speech by Modeling the Active Voice Source
Proceedings of International Conference on Signals and Electronic Systems, 2006
Learning for Bone-Conducted Speech via Adaptive and Neural Filters
Proceedings of International Conference on Signals and Electronic Systems, 2006
Pitch Determination using Aligned AMDF
Proceedings of International Conference on Spoken Language Processing, 2006
A Harsh Noise Assessment Measure for Speech Enhancement
Proceedings of European Conference on Signal Processing, 2006
Improving Bone-Conducted Speech Quality via Neural Network
Proceedings of IEEE International Symposium on Signal Processing and Information Technology, 2006
SIDOモデルを用いたブラインド等化に関する一検討
電子情報通信学会技術研究報告, 2006
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, 2006
Reconstruction Filter design for Bone-Conducted Speech
Proceedings of International Conference on Spoken Language Processing, 2006
骨導音声の品質改善について （その１）
日本音響学会2005年春季研究発表会講演論文集, 2006
Poster presentation
反復RFアルゴリズムを利用したリアルタイム騒音制御
日本音響学会春季研究発表会講演論文集, 2006
Poster presentation
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
日本音響学会春季研究発表会講演論文集, 2006
線形予測フィルタを用いた適応音声強調
日本音響学会春季研究発表会講演論文集, 2006
Poster presentation
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
日本音響学会春季研究発表会講演論文集, 2006
Poster presentation
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
日本音響学会春季研究発表会講演論文集, 2006
Poster presentation
A Study on Normalized LMS Algorithm Using Refined Filtering Technique
Proceedings of 5th WSEAS International Conference on Signal Processing, Robotics and Automation, 2006
Noise Estimation Using Multifrequency Regions for Spectral Subtraction
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Adaptive Time-Variant Channel Equalization Using Phase-Banded LMS Algorithm
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Active Noise Control Using Cascaded Adaptive Filters
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2006
Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
2005
A Parallel Estimator with LMS Adaptation for Fast Fading Channels
2005
Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
2005
Speech Enhancement Using a Technique of Adaptive Bias Suppression
2005
Poster presentation
Voice Source Modeling for Accurate Speech Analysis
2005
Quality Improvement of Bone-Conducted Speech
2005
A Reconstruction Filter for Bone-Conducted Speech
2005
Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method
2005
Poster presentation
Linear Prediction Using Homomorphic Deconvolution in the Autocorrelation Domain
2005
Poster presentation
Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
2005
Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
2005
LMS-VSLMS縦列接続による適応非線形予測分析
2005
Poster presentation
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
2005
Poster presentation
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
2005
Poster presentation
適応バイアス抑制技術を用いた音声強調
2005
骨導音声の品質改善について （その２）
2005
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
2005
White Noise Removal in Image by Iterative Spectral Subtraction Method
2005
Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
2005
Equalization of Time Variant Multipath Channels Using Channel Estimation Based Classification Techniques
Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology, 2005
A Parallel Estimator with LMS Adaptation for Fast Fading Channels
Proceedings of IEEE Region 10 International Conference on Electrical and Electronic Technology, 2005
Spectral Subtraction with Non-Stationary Noise Estimation Utilizing Harmonic Structure
Proceedings of WSEAS International Conference on Electronics, Control and Signal Processing, 2005
Speech Enhancement Using a Technique of Adaptive Bias Suppression
Proceedings of Asilomar Conference on Signals, Systems and Computers, 2005
Poster presentation
Voice Source Modeling for Accurate Speech Analysis
Proceedings of Asilomar Conference on Signals, Systems and Computers, 2005
Quality Improvement of Bone-Conducted Speech
Proceedings of European Conference on Circuit Theory and Design, 2005
A Reconstruction Filter for Bone-Conducted Speech
Proceedings of IEEE International Midwest Symposium on Circuits and Systems, 2005
Restoration from Image Degraded by White Noise Based on Iterative Spectral Subtraction Method
Proceedings of IEEE International Symposium on Circuits and Systems, 2005
Poster presentation
Linear Prediction Using Homomorphic Deconvolution in the Autocorrelation Domain
Proceedings of IEEE International Symposium on Circuits and Systems, 2005
Poster presentation
Adaptive Nonlinear Predictive Analysis for Speech Using a Cascaded LMS-VSLMS Predictor
Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing, 2005
Sinusoidal Time Delay Tracking by a Self-tuned LMS Filter with Interpolation Based on System Response Coefficient Ratios
Proceedings of IEEE-EURASIP Workshop on Nonlinear Signal and Image Processing, 2005
LMS-VSLMS縦列接続による適応非線形予測分析
日本音響学会2005年春季研究発表会講演論文集, 2005
Poster presentation
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2005年春季研究発表会講演論文集, 2005
Poster presentation
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2005年春季研究発表会講演論文集, 2005
Poster presentation
適応バイアス抑制技術を用いた音声強調
日本音響学会2005年春季研究発表会講演論文集, 2005
骨導音声の品質改善について （その２）
日本音響学会2005年春季研究発表会講演論文集, 2005
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2005年春季研究発表会講演論文集, 2005
White Noise Removal in Image by Iterative Spectral Subtraction Method
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2005
Noise Removal for Image Degraded by Poisson Noise: A Pixel Values Based Division Approach
Proceedings of RISP International Workshop on Nonlinear Circuits and Signal Processing, 2005
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-
2004
Nonlinear Predictive Analysis of Speech by Iterative Approach
2004
Pitch Synchronous Addition and Extension for Linear Predictive Analysis of Noisy Speech
2004
Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
2004
Poster presentation
Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
2004
Adaptive Non-linear Prediction of Speech in Impulse Noise
2004
Poster presentation
Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
2004
Poster presentation
A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
2004
Equalizer-aided time delay tracking based on finite differences
2004
An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
2004
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, 2004
Nonlinear Predictive Analysis of Speech by Iterative Approach
Proc. 12th Europian Signal Processing Conf., 2004
Pitch Synchronous Addition and Extension for Linear Predictive Analysis of Noisy Speech
Proc. 6th Nordic Signal Processing Symposium, 2004
Non-Stationary Noise Estimation Utilizing Harmonic Structure for Spectral Subtraction
Proceedings of Asilomar Conference on Signals, Systems and Computers, 2004
Poster presentation
Noise-Robust Fundamental Frequency Extraction Method Based on Exponentiated Band-Limited Amplitude Spectrum
Proc. 47th IEEE International Midwest Symposium on Circuits and Systems, 2004
Adaptive Non-linear Prediction of Speech in Impulse Noise
Proc. 18th International Congress on Acoustics, 2004
Poster presentation
Improved model of SPAC (Speech Processing System by use of autocorrelation function) utilizing spectral subtraction as preprocessing
Proc. 18th International Congress on Acoustics, 2004
Poster presentation
A Novel Amplitude Banded RLS Algorithm for Equalization of Time Variant Multipath Channels
Proceedings of IFAC Workshop on Adaptation and Learning in Control and Signal Processing, 2004
Equalizer-aided time delay tracking based on finite differences
Proc. 19th IEICE Signal Processing Symposium, 2004
An adaptive Butler-Cantoni based time delay estimation (ABCTDE) method- IIR whitening filter approach
Proceedings of IEEE International Symposium on Circuits and Systems, 2004
