Yoichi HANEDA
| Department of Informatics | Professor |
| Cluster I (Informatics and Computer Engineering) | Professor |
| Fundamental Program for Advanced Engineering (Evening Course) | Professor |
Researcher Information
Research Keyword
Field Of Study
Career
- 01 Apr. 2016
電気通信大学, 大学院 情報理工学研究科 情報学専攻, 教授 - 01 Oct. 2012 - 31 Mar. 2016
電気通信大学大学院 情報理工学研究科 総合情報学専攻, 教授 - 01 Apr. 2008 - 30 Sep. 2012
日本電信電話（株）NTTサイバースペース研究所, グループリーダー - 01 Oct. 2002 - 31 Mar. 2008
日本電信電話（株）NTTサイバースペース研究所, 主幹研究員 - 01 Apr. 2000 - 30 Sep. 2002
日本電信電話（株）NTTサイバースペース研究所, 主任研究員 - 01 Apr. 1989 - 31 Mar. 1998
日本電信電話（株）NTTヒューマンインタフェース研究所
Educational Background
Member History
- May 2023
President, Acoustical society of Japan, Society - 01 Jun. 2017 - 31 May 2019
副会長, 日本音響学会, Society - 01 Jun. 2015 - 31 May 2019
理事, 日本音響学会, Society - 01 Jun. 2015 - 31 May 2017
出版委員長, 日本音響学会, Society - 10 Mar. 2014 - 31 May 2017
音響入門シリーズ編集委員長, 日本音響学会, Society - 01 Jun. 2016 - 30 May 2017
応用音響研究会(EA) 副委員長, 電子情報通信学会, Society - 01 Jun. 2015 - 31 May 2016
応用音響研究会委員長, 電子情報通信学会, Society - 01 Jun. 2014 - 30 May 2015
応用音響研究会(EA) 副委員長, 電子情報通信学会, Society - Jun. 2013 - May 2015
音響学入門ペディア作成委員会委員長, 日本音響学会, Society - 14 Jul. 2014
音響WGサブリーダー, 超臨場感コミュニケーション産学官フォーラム, Others - May 2009 - May 2013
理事, 日本音響学会, Society - May 2002 - May 2012
Audio and Acoustic Signal Processing Technical Committee, Society - May 2007 - May 2009
基礎・境界ソサイエティ 事業担当幹事, 電子情報通信学会, Society - May 2007 - May 2009
広報委員会 委員, 電子情報通信学会, Society - May 2007 - May 2009
会員増強委員会委員, 電子情報通信学会, Society - May 2003
評議員, 日本音響学会, Society
Research Activity Information
Award
- Oct. 2023
IEEE Global Conference on Consumer Electronics
Investigation of Loudspeaker Array Layout for Sound Field Reproduction by Room-Mode Matching in Rectangular Rooms
IEEE GCCE 2023 Excellent Paper Awards (Bronze Prize), Natsuko Maeda;Yoichi Haneda - Mar. 2019
RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing 2016
Directivity control using two circular loudspeaker arrays
Student Paper Award in NCSP'19, Yi Ren
International society - Mar. 2018
日本音響学会
円調和・縦型多重極表現に基づく円筒形アレイの3次元指向性制御
日本音響学会学生優秀発表賞, 佐藤航也
Japan society - Oct. 2017
IEEE
L1-Regularised MV Beamformer in the Spherical Harmonic Domain for Spherical Microphone Array Recording
2017 IEEE 6th Global Conference on Consumer Electronics Outstanding Poster Award, Yuhei Yamamoto and Yoichi Haneda
International society - Oct. 2017
IEEE
2017 IEEE 6th Global Conference on Consumer Electronics Outstanding Poster Award, Yuhei Yamamoto;Yoichi Haneda
International society - Mar. 2017
日本音響学会
聴感評点を最大化するための強化学習に基づく音源強調の検討
日本音響学会粟屋潔学術奨励賞, 小泉悠馬
Japan society - Dec. 2016
Acoustical Society of America
America
Best paper by a young presenter in signal processing in acoustics, Maureen
International society, United States - Mar. 2016
RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing 2016
Student Paper Award in NCSP'16, Kazuna Bando
International society - Mar. 2015
電気通信普及財団
第３０回電気通信普及財団賞（テレコムシステム技術省）, 小山翔一,古家賢一,日和崎祐介,羽田陽一
Publisher - Mar. 2015
電気通信普及財団
第３０回電気通信普及財団賞（テレコムシステム技術省）, 小山翔一;古家賢一;日和崎祐介;羽田陽一
Publisher - May 2007
日本音響学会
日本音響学会 技術開発賞 - Mar. 2005
電子情報通信学会
電子情報通信学会ヒューマンコミュニケーション賞 - May 2002
電子情報通信学会
電子情報通信学会 論文賞 - Mar. 2002
日本音響学会
日本音響学会 佐藤論文賞 - Mar. 1998
日本音響学会
日本音響学会 粟屋潔学術奨励賞 - May 1997
電子情報通信学会 第34回 業績賞 - Mar. 1996
電子情報通信学会
電子情報通信学会 学術奨励賞 - May 1995
日本音響学会
日本音響学会 技術開発賞
Paper
- Subjective Evaluation of a Focused Sound Source Reproducing at the positions of a Listener’s Moving Hand
Miho Hirohashi; Yoichi Haneda
Last, 2023 Asia Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC), IEEE, 2395-2401, 31 Oct. 2023, Peer-reviwed
International conference proceedings, English - Investigation of Loudspeaker Array Layout for Sound Field Reproduction by Room-Mode Matching in Rectangular Rooms
Natsuko Maeda; Yoichi Haneda
Last, Proceeding of GCCE 2023, Oct. 2023, Peer-reviwed
International conference proceedings, English - Sound field control in a rectangular closed space using multiple selected room modes
Natsuko Maeda; Yoichi Haneda
Last, Proceeding of Inter Noise 2023, Aug. 2023, Peer-reviwed
International conference proceedings, English - Reproducing various wavefronts at low frequencies in a rectangular closed space using room-mode matching with DCT
Natsuko Maeda; Yoichi Haneda
Last, Audio Engineering Society Convention, 154, 13 May 2023, Peer-reviwed
International conference proceedings, English - Focused Source Reproduction using Rigid Elliptical Loudspeaker Array
Yi Ren and Yoichi Haneda
Proceedings of ICA (International Congress of Acoustics) 2022, A13, 05, ABS-008233, 26 Oct. 2022, Peer-reviwed, False
International conference proceedings, English - Ensemble Learning Approach With Class Rotation for Three-Dimensional Classification on Direction-of-Arrival Estimation
Israel Mendoza-Velazquez; Hector Perez-Meana; Yoichi Haneda
IEEE Access, IEEE, 10, 108185-108193, 20 Oct. 2022, Peer-reviwed
Scientific journal, English - Evaluation of Moving Sound Image Rendering by VBAP and HOA With Distance Gain Control
Kensuke Kubo; Yoichi Haneda
IEEE GCCE 2022, IEEE, POS, 1, 7-8, 17 Oct. 2022, Peer-reviwed
International conference proceedings, English - Two-dimensional sound field reproduction based on Mathieu function expansion
Yi Ren; Yoichi Haneda
The Journal of the Acoustical Society of America, Acoustical Society of America, 152, 1, 416-428, 28 Jul. 2022, Peer-reviwed
Scientific journal, English - Spherical Convolutional Recurrent Neural Network for Real-Time Sound Source Tracking
Tianle Zhong; Israel Mendoza Velázquez; Yi Ren; Héctor Manuel; Pérez Meana; Yoichi Haneda
IEEE International Conference on Acoustics, Speech and Signal Processing 2022(ICASSP 2022), IEEE, 5063-5067, 23 May 2022, Peer-reviwed
International conference proceedings, English - 2D Local Exterior Sound Field Reproduction Using an Addition Theorem Based on Circular Harmonic Expansion
Yi Ren; Yoichi Haneda
2021 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), IEEE, 271-275, 17 Oct. 2021, Peer-reviwed
International conference proceedings, English - Involution Based Speech Autoencoder: Investigating the Advanced Vision Operator Performance on Speech Feature Extraction
Tianle Zhong; Israel Velázquez; Yoichi Haneda
2021 IEEE 10th Global Conference on Consumer Electronics (GCCE 2021), IEEE, OS-NLP(2), 12 Oct. 2021, Peer-reviwed
International conference proceedings, English - DOA Estimation for Spherical Microphone Array Using Spherical Convolutional Neural Networks Extraction
Israel Velázquez; Yi Ren; Yoichi Haneda
2021 IEEE 10th Global Conference on Consumer Electronics (GCCE 2021), IEEE, OS-AAT(2), 12 Oct. 2021, Peer-reviwed
International conference proceedings, English - A FUSION METHOD BASED ON CLASS ROTATIONS FOR DNN-DOA ESTIMATION ON SPHERICAL MICROPHONE ARRAY
Israel Mendoza Velázquez; Yi Ren; Yoichi Haneda; Héctor Manuel; Pérez Meana
EUSIPCO, 6506, 24 Aug. 2021, Peer-reviwed
International conference proceedings, English - Beamforming using two rigid circular loudspeaker arrays: Numerical simulations and experiments
Yi Ren; Yoichi Haneda
Audio Engineering Society Convention 150, 10450, 24 May 2021
International conference proceedings, English - Two-dimensional exterior sound field reproduction using two rigid circular loudspeaker arrays
Yi Ren; Yoichi Haneda
The Journal of Acoustical Society of America, The Acoustical Society of America, 148, 4, 2236-2247, 20 Oct. 2020, Peer-reviwed
Scientific journal, English - 2D Sound Field Reproduction with Elliptical Loudspeaker Array based on Circular Microphone Array Signals
Yi Ren; Kenta Imaizumi; Kimitaka Tsutsumi; Yoichi Haneda
Audio Engineering Society Convention 148, 143, 10327, 28 May 2020, Peer-reviwed
International conference proceedings, English - Discrimination method of direction of arrival estimation correctness based on deep neural network
Ryusuke Tanaka; Yoichi Haneda
Acoustical Science and Technology, ACOUSTICAL SOCIETY OF JAPAN, 41, 1, 318-321, 01 Jan. 2020, Peer-reviwed
Scientific journal, English - Sound field synthesis based on superposition of multipoles comprising focused monopole sources
Tsutsumi Kimitaka; Imaizumi Kenta; Haneda Yoichi; Takada Hideaki
Acoustical Science and Technology, ACOUSTICAL SOCIETY OF JAPAN, 41, 2, 489-500, 2020, Peer-reviwed,We propose a method to create a directional sound source in front of a linear loudspeaker array. The method creates clusters of focused sources to form multipoles by using a linear loudspeaker array and superposes the multipoles to synthesize a directivity pattern. We also derive an efficient multipole structure in which adjacent lower order multipoles are overlapped. The structure reduces the number of focused sources, thereby reducing the algorithmic complexity needed to create them. To further reduce complexity, we also derive a time domain implementation of the proposed method. To mitigate degradation in the reproduced directivity due to superposition of the inaccurate sound fields of focused sources, a fractional delay interpolation is applied. Computer simulation results indicate that the proposed method based on superposition of up to the third order multipoles creates a directional sound source at significantly lower complexity than a conventional method.
English - Analytical Method of 2.5 d Exterior Sound Field Synthesis By Using Multipole Loudspeaker Array
Kenta Imaizumi; Kimitaka Tsutsumi; Atsushi Nakadaira; Yoichi Haneda
2019 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), IEEE, 388-392, 20 Oct. 2019, Peer-reviwed
International conference proceedings, English - How the distance and radius of two circular loudspeaker arrays affect sound field reproductions and directivity controls
Yi Ren; Yoichi Haneda
23RD INTERNATIONAL CONGRESS ON ACOUSTICS, 1140-1147, 09 Sep. 2019, Peer-reviwed
International conference proceedings, English - Investigation into Transaural System with Beamforming Using a Circular Loudspeaker Array set at Off-center Position from the Listener
Yu Ito; Yoichi Haneda
23RD INTERNATIONAL CONGRESS ON ACOUSTICS, 2773-2780, 09 Sep. 2019, Peer-reviwed
International conference proceedings, English - Analytical method to convert circular harmonic expansion coefficients for sound field synthesis by using multipole loudspeaker array
Kimitaka Tsutsumi; Kenta Imaizumi; Atsushi Nakadaira; Yoichi Haneda
2019 27th European Signal Processing Conference (EUSIPCO), 1-5, 02 Sep. 2019, Peer-reviwed
International conference proceedings, English - Directivity Control Using Two Circular Loudspeaker Arrays
Y Ren; Y Haneda
Journal of Signal Processing, 23, 4, 159-162, 20 Jul. 2019, Peer-reviwed
Scientific journal, English - Trainable Adaptive Window Switching for Speech Enhancement
Yuma Koizumi; Noboru Harada; Yoichi Haneda
2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 616-620, 12 May 2019, Peer-reviwed
International conference proceedings, English - Directivity control of a finite cylindrical loudspeaker array based on circular harmonics and longitudinal multipole expression
K Sato; Y Haneda
Acoustical Science and Technology, 40, 2, 93-104, 01 Mar. 2019, Peer-reviwed
Scientific journal, English - Virtual Source Reproduction Using Two Rigid Circular Loudspeaker ArraysLoudspeaker Array
Y. Ren; Y. Haneda
Audio Engineering Society 145th Convention, 145, 10120, 07 Oct. 2018, Peer-reviwed
International conference proceedings, English - Horizontal Binaural Signal Generation at Semi-Arbitrary Positions Using a Linear Microphone Array
A. Yamazato; Y. Haneda
Audio Engineering Society 145th Convention, 145, 10122, 07 Oct. 2018, Peer-reviwed
International conference proceedings, English - MUSIC ALGORITHM BASED ON TEMPORAL DRR AND RAYLEIGH QUOTIENT FOR REVERBERANT ENVIRONMENTS
R. Tanaka; Y. Haneda
International Workshop on Acoustic Signal Enhancement 2018 (IWAENC 2018), P3, 20, 17 Sep. 2018, Peer-reviwed
International conference proceedings, English - DNN-based Source Enhancement to Increase Objective Sound Quality Assessment Score
Y. Koizumi; K. Niwa; Y. Hioka; K. Koabayashi; Y. Haneda
IEEE/ACM Trans. Audio, Speech, and Lang. Process., IEEE, 26, 10, 1780-1792, 30 May 2018, Peer-reviwed
Scientific journal, English - Sound Localization of Beamforming-Controlled Reflected Sound from Ceiling in Presence of Direct Sound
H. Sakamoto; Y. Haneda
Audio Engineering Society 144th Convention, 144, 9949, 14 May 2018, Peer-reviwed
International conference proceedings, English - END-TO-END SOUND SOURCE ENHANCEMENT USING DEEP NEURAL NETWORK IN THE MODIFIED DISCRETE COSINE TRANSFORM DOMAIN
Y. Koizumi; N. Harada; Y. Haneda; Y. Hioka; K. Kobayashi
2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2018, 1274, 15 Apr. 2018, Peer-reviwed
International conference proceedings, English - DIRECTIVITY SYNTHESIS WITH MULTIPOLES COMPRISING A CLUSTER OF FOCUSED SOURCES USING A LINEAR LOUDSPEAKER ARRAY
Kimitaka Tsutsumi; Yoichi Haneda; Ken'ichi Noguchi; Hideaki Takada
2018 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 2018, 496-500, 2018, Peer-reviwed, A method to create multipoles comprising a cluster of focused sources by using a linear loudspeaker array has recently been investigated. Directivities in a listening area were confirmed with examples of primitive multipoles such as dipoles and quadrupoles. This paper describes a method to create a sound source having more complex directivity by using a superposition of multipoles comprising a collection of focused sources. An analytical method is also described with which coefficients can be obtained for each multipole by circular harmonic expansion of a sound field created by a directional sound source and Taylor expansion of the corresponding sound field. Simulation results show that a superposition of multipoles based on analytical conversion introduces desired directivities to the sound sources created in the listening area by a linear loudspeaker array.
International conference proceedings, English - Sidelobe suppression by desired directivity pattern optimization for a small circular loudspeaker array
Koya Sato; Yoichi Haneda
Acoustical Science and Technology, Acoustical Society of Japan, 39, 3, 243-251, 2018, Peer-reviwed, Directivity control using a loudspeaker array is widely studied for various applications. Suppressing sidelobe levels is important for applications such as personal audio systems. In this paper, we propose a filter design method using a window function shape as the desired directivity pattern to reduce the sidelobe levels. The proposed method consists of three steps. The first step defines a cost function with a criterion for the directivity pattern. Next, filter coefficients for each loudspeaker are calculated and stored while changing the window function shape of the desired directivity pattern. Finally, we determine the optimum filter coefficients having the best performance of the cost function by using a full-search algorithm at each frequency. We conducted directivity experiments with a real six-element circular loudspeaker array having a radius of 0.055 m and evaluated its directivity to confirm the performance of the proposed method. The results, which were compared with those obtained from a conventional method, showed that the maximum sidelobe level improved by about 2 dB, although the beam was wide. We verified that using the window function shape as the desired directivity pattern is more effective than using the conventional method for sidelobe suppression.
Scientific journal, English - A method to reproduce a directional sound source by using a circular array of focused sources in front of a linear loudspeaker array
Kimitaka Tsutsumi; Yoichi Haneda; Kenichi Noguchi; Hideaki Takada
144th Audio Engineering Society Convention 2018, 144, 9978, 2018, Peer-reviwed, © 2018 Audio Engineering Society. All Rights Reserved. We propose a method to create a directional sound source in front of a linear loudspeaker array. The method creates a virtual circular loudspeaker array comprising multiple focused sources to reproduce directivity patterns. In the proposed method, the driving functions for the secondary sources are defined as a cascade combination of two driving functions: The first one for directivity control derived by an analytical conversion of circular harmonic modes, and the second one for creating focused sources. The proposed driving functions can deal with directivity rotation by changing the position of focused sources, thereby avoiding recalculations of driving functions. Using computer simulation, we obtained accuracy and algorithmic complexity comparable or better than those of a conventional method.
International conference proceedings, English - 3D directivity control of a finite cylindrical loudspeaker array with cylindrical harmonics
Koya Sato; Yoichi Haneda
2017 IEEE 6th Global Conference on Consumer Electronics, GCCE 2017, Institute of Electrical and Electronics Engineers Inc., 2017-, 2017, 1-2, 19 Dec. 2017, Peer-reviwed, We propose a filter design method based on three-dimensional (3D) bases for a finite cylindrical loudspeaker array. In this method, we first derive the relationship between the 3D bases of directivity patterns and finite cylindrical harmonics. Directivity simulations were conducted using a prototypical 24-element cylindrical array with a radius of 0.07 m and height of 0.28 m and compared with the results of an 8-element circular array. These results showed that the cylindrical array can suppress sound pressure level of approximately 10 dB in the z-axis direction although the x-y plane was similar in comparison to the circular array. Moreover, we confirmed that the cylindrical array can control directivity in a diagonal direction.
International conference proceedings, English - L1-regularised MV beamformer in the spherical harmonic domain for spherical microphone array recording
Yuhei Yamamoto; Yoichi Haneda
2017 IEEE 6th Global Conference on Consumer Electronics, GCCE 2017, Institute of Electrical and Electronics Engineers Inc., 2017-, 2017, 1-2, 19 Dec. 2017, Peer-reviwed, Teleconference systems are widely used in business and education. They are required to enhance only the target speech while suppressing diffuse or interference noise. The minimum variance beamformer is a strong method to enhance the target speech
However, it is less robust at low frequencies. We propose a robust beamformer based on the minimum variance method in the spherical harmonic domain. Less robustness is caused by the use of high-order spherical harmonic coefficients at low frequencies. To avoid the use of high-order coefficients in the filter calculation, we apply L1-regularization to the minimum variance method. Our analysis demonstrates the tradeoff between the robustness and spatial resolution and how the L1-regularization improves the robustness at low frequencies.
International conference proceedings, English - Filter Design of a Circular Loudspeaker Array Considering the Three-Dimensional Directivity Patterns Reproduced by Circular Harmonic Modes
K. Sato; Y. Haneda
Audio Engineering Society 142nd Convention, 142, 9767, 11 May 2017, Peer-reviwed
International conference proceedings, English - Wearable Sound Reproduction System Using Two End-Fire Arrays
K. Imaizumi; Y. Haneda
Audio Engineering Society 142nd Convention, 142, 9768, 11 May 2017, Peer-reviwed
International conference proceedings, English - DNN-BASED SOURCE ENHANCEMENT SELF-OPTIMIZED BY REINFORCEMENT LEARNING USING SOUND QUALITY MEASUREMENTS
Yuma Koizumi; Kenta Niwa; Yusuke Hioka; Kazunori Kobayashi; Yoichi Haneda
2017 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 2017, 81-85, 2017, We investigated whether a deep neural network (DNN)-based source enhancement function can be self-optimized by reinforcement learning (RL). The use of a DNN is a powerful approach to describing the relationship between two sets of variables and can be useful for source enhancement function design. By training the DNN using a huge amount of training data, sound quality of output signals are improved. However, collecting a huge amount of training data is often difficult in practice. To use limited training data efficiently, we focus on the "self-optimization" of DNN-based source enhancement function in which RL is commonly utilized in the development of game playing computers. As a reward for RL, quantitative metrics that reflect a human's perceptual score (perceptual score), e.g., perceptual evaluation methods for audio source separation (PEASS), are utilized. To investigate whether the sound quality is improved by RL-based source enhancement, subjective tests were conducted. It was confirmed that the output sound quality of the RL-based source enhancement function improved as the number of iterations was increased and finally outperformed the conventional method.
International conference proceedings, English - Generating method of binaural signals at semi- arbitrary listening positions from a linear microphone array
A. Yamazato; Y. Haneda
EA2017-80, 電子情報通信学会, 117, 328,, 115-120, 2017
Symposium, English - Visualization of one source in a two sources sound field by estimating source positions
Maureen; Y. Haneda
Joint Meeting of the Acoustical Society of America and Acoustical Society of Japan, Acoustical Society of America, 140, 4, 3059-3059, 28 Nov. 2016, Peer-reviwed
International conference proceedings, English - Directivity control for regular polyhedron loudspeaker array based on weighted least-squares method using adjusted weight in spherical harmonics domain
M. Daikohara; Y. Haneda
Joint Meeting of the Acoustical Society of America and Acoustical Society of Japan, 140, 4, 3056-3056, 28 Nov. 2016, Peer-reviwed
International conference proceedings, English - Personal audio reproduction using two wearable end-fire loudspeaker arrays
K. Imaizumi; Y. Haneda
Joint Meeting of the Acoustical Society of America and Acoustical Society of Japan, Acoustical Society of America, 140, 4, 3057-3057, 28 Nov. 2016, Peer-reviwed
International conference proceedings, English - Directivity control of a compact circular loudspeaker array based on selected orders of circular harmonic expansion
K. Sato; Y. Haneda
Joint Meeting of the Acoustical Society of America and Acoustical Society of Japan, Acoustical Society of America, 140, 4, 3061-3061, 28 Nov. 2016, Peer-reviwed
International conference proceedings, English - Acoustic Echo and Noise Canceller for Personal Hands-Free Video IP Phone
Masahiro Fukui; Suehiro Shimauchi; Yusuke Hioka; Akira Nakagawa; Yoichi Haneda
IEEE TRANSACTIONS ON CONSUMER ELECTRONICS, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 62, 4, 454-462, Nov. 2016, Peer-reviwed, This paper presents implementation and evaluation of a proposed acoustic echo and noise canceller (AENC) for videotelephony-enabled personal hands-free internet protocol (IP) phones. This canceller has the following features: noise-robust performance, low processing delay, and low computational complexity. The AENC employs an adaptive digital filter (ADF) and noise reduction (NR) methods that can effectively eliminate undesired acoustic echo and background noise included in a microphone signal even in a noisy environment. The ADF method uses the step-size control approach according to the level of disturbance such as background noise; it can minimize the effect of disturbance in a noisy environment. The NR method estimates the noise level under an assumption that the noise amplitude spectrum is constant in a short period, which cannot be applied to the amplitude spectrum of speech. In addition, this paper presents the method for decreasing the computational complexity of the ADF process without increasing the processing delay to make the processing suitable for real-time implementation. The experimental results demonstrate that the proposed AENC suppresses echo and noise sufficiently in a noisy environment; thus, resulting in natural-sounding speech(1).
Scientific journal, English - Interactive Directivity Control Using Dodecahedron Loudspeaker Array
K. Bando; Y. Haneda
Journal of Signal Processing, Journal of Signal Processing, 20, 4, 209-212, 01 Jul. 2016, Peer-reviwed, In this paper, we investigate loudspeaker array directivity control and its application to sound localization when steering the beam direction of the loudspeaker array. The filter of a spherical loudspeaker array is designed based on the minimum variance (MV) method [1] in the spherical harmonics (SH) domain [2]. The beam pattern generated by the filter achieves a high directivity in a certain direction. Further, the beam direction can be rotated while maintaining the shape of the beam pattern. We implemented a prototype system that allows the listener to steer the beam direction using a gyro sensor in real time. The listening test result suggested that the location of the sound image varies based on the sound reflected from the wall. We evaluated the sound localization mechanism while rotating the beam direction based on the observed signal measured with a dummy head. The result showed that the sound of the beam reflected from a wall influences horizontal plane localization and sound image distance.
Scientific journal, English - Interactive directivity control using dodecahedron loudspeaker array
K. Bando; Y. Haneda
The proceddings of the 2016 RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing, Research Institute of Signal Processing, Japan, 20, 4, 9-12, 07 Mar. 2016, Peer-reviwed, In this paper, we investigate loudspeaker array directivity control and its application to sound localization when steering the beam direction of the loudspeaker array. The filter of a spherical loudspeaker array is designed based on the minimum variance (MV) method [1] in the spherical harmonics (SH) domain [2]. The beam pattern generated by the filter achieves a high directivity in a certain direction. Further, the beam direction can be rotated while maintaining the shape of the beam pattern. We implemented a prototype system that allows the listener to steer the beam direction using a gyro sensor in real time. The listening test result suggested that the location of the sound image varies based on the sound reflected from the wall. We evaluated the sound localization mechanism while rotating the beam direction based on the observed signal measured with a dummy head. The result showed that the sound of the beam reflected from a wall influences horizontal plane localization and sound image distance.
International conference proceedings, English - SPHERICAL MICROPHONE ARRAY POST-FILTERING FOR REVERBERATION SUPPRESSION USING ISOTROPIC BEAMFORMINGS
Yuhei Yamamoto; Yoichi Haneda
2016 IEEE INTERNATIONAL WORKSHOP ON ACOUSTIC SIGNAL ENHANCEMENT (IWAENC), IEEE, PS-I, 17, 2016, Peer-reviwed, Dereverberation or reverberation suppression techniques are important for many applications. Beamforming is an effective technique for speech enhancement; however, it is impossible to suppress the reverberation arriving from the target direction for beamforming. The non-linear post-filtering methods have been proposed for reverberation suppression based on Wiener filtering. Post-filtering can suppress the reverberation from the output of beamforming when the power spectral density (PSD) of the desired reverberation is estimated. In this paper, we assume that reverberation arrives isotropically. Under this assumption, using isotropic beamforming on the spherical harmonic (SH) domain, the PSD of reverberation can be estimated. Based on our experiments, the proposed method is an effective tool for suppressing reverberation; furthermore, it can be used to reduce background noise.
International conference proceedings, English - DIRECTION-OF-ARRIVAL ESTIMATION BASED ON JOINT DIAGONALIZATION OF MATRICES IN DIFFERENT DIRECT-TO-REVERBERATION RATIOS
Ryusuke Tanaka; Yoichi Haneda
2016 IEEE INTERNATIONAL WORKSHOP ON ACOUSTIC SIGNAL ENHANCEMENT (IWAENC), IEEE, PS-I, 19, 2016, Peer-reviwed, Direction-of-arrival (DOA) estimation is very important in many applications. The MUltiple SIgnal Classification (MUSIC) algorithm is one of the well-known methods of DOA estimation. However, in a real environment with reverberation, MUSIC's estimation accuracy is degraded. In this paper, we propose an improved MUSIC algorithm by using joint diagonalization of the covariance matrices of two different direct-to-reverberation ratio (DRR) periods corresponding to higher and lower DRR periods in a short time. In addition, we also introduce a method for detecting different DRR periods by using peak-hold processing for our proposed method. Computer simulations verified that the proposed method improves the accuracy of the azimuth in DOA estimation compared with the conventional method for both simulated and actual impulse responses.
International conference proceedings, English - Linear array aperture extrapolation based on a spherical wave assumption
Maureen; Y. Haneda
12th Western Pacific Acoustics Conference 2015, 393-396, 03 Dec. 2015, Peer-reviwed
International conference proceedings, English - 高品質IP電話トライアル向け広帯域ハンドセットの開発
岡本学; 野口賢一; 日和崎祐介; 羽田陽一
日本音響学会誌, The Acoustical Society of Japan (ASJ), 71, 11, 581-589, 01 Nov. 2015, Peer-reviwed, 2008年からNTTグループがサービスを開始したNGNでは高音質電話サービスが提供されている。サービス提供前に行われたフィールドトライアルでは,サービスの有効性を検証するために高品質IP電話機を試作し,それに合わせ広帯域ハンドセットも開発した。開発では,まず目標特性の設定を行い,続けてマイクロホン,イヤホンの構造等の変更により目標特性を達成した。開発したハンドセット及び高品質IP電話機を用い評価を行った結果,従来のハンドセットに比べ十分に高品質であること,加えてフィールドトライアルにおいてサービスの有効性を確認できた。
Scientific journal, Japanese - Source-Location-Informed Sound Field Recording and Reproduction
Shoichi Koyama; Ken'ichi Furuya; Yoichi Haneda; Hiroshi Saruwatari
IEEE JOURNAL OF SELECTED TOPICS IN SIGNAL PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 9, 5, 881-894, Aug. 2015, Peer-reviwed, A sound field recording and reproduction method that exploits prior information on the locations of sound sources to be reproduced is proposed. Current methods do not take such prior information into consideration in the transformation from the signals received by microphones into the driving signals of loudspeakers. The proposed method for planar and linear arrays of microphones and loudspeakers optimizes spatial basis functions and their coefficients, which represent the driving signals of loudspeakers, on the basis of the maximum a posteriori (MAP) estimation approach, leading to higher reproduction accuracy above the spatial Nyquist frequency when there are fewer microphones than loudspeakers, i.e., super-resolution in sound field recording and reproduction. Numerical simulations show that the reproduction accuracy at frequencies above the spatial Nyquist frequency was maintained using the proposed method even when there was a mismatch between actual and prior locations of the sound sources to be reproduced.
Scientific journal, English - SOURCE-LOCATION-INFORMED SOUND FIELD RECORDING AND REPRODUCTION WITH SPHERICAL ARRAYS
Shoichi Koyama; Koichiro Ito; Hiroshi Saruwatari
2015 IEEE WORKSHOP ON APPLICATIONS OF SIGNAL PROCESSING TO AUDIO AND ACOUSTICS (WASPAA), IEEE, 9, 5, 881-894, 2015, Peer-reviwed, A sound field recording and reproduction method for spherical arrays that takes into account prior information on the locations of sound sources to be reproduced, i.e., primary sources, is proposed. Current methods using spherical arrays of microphones and loudspeakers, such as higher order ambisonics (HOA), are based on sound field analysis and synthesis in a spherical harmonic domain. However, the size of the reproduced region is limited by the maximum available order of the spherical harmonics, which is usually determined by the microphone array design. Our proposed method exploits approximate positions of primary sources, which may be estimated from the signals received by microphones or other sensor data, to obtain the driving signals of the loudspeakers. Spatial basis functions and their coefficients representing the driving signals are optimized on the basis of maximum a posteriori (MAP) estimation using the prior source locations. Numerical simulation results indicated that the reproduced region can be enlarged to greater than that when using HOA by using the proposed method.
International conference proceedings, English - Wave Field Reconstruction Filtering in Cylindrical Harmonic Domain for With-Height Recording and Reproduction
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda; Yoiti Suzuki
IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 22, 10, 1546-1557, Oct. 2014, Peer-reviwed, For sound field reproduction that includes height (with-height reproduction), it is more efficient to record and reproduce the sound field with lower resolution in elevation than in azimuth due to the spatial abilities of human auditory perception. We propose a sound field reproduction method using horizontally arranged cylindrical arrays of microphones and loudspeakers, which is based on the wave field reconstruction (WFR) filter. With the use of cylindrical array configurations, it is possible to reproduce sound waves arriving from upper and lower directions with a smaller number of array elements at angular positions. The WFR filter is analytically derived in the cylindrical harmonic domain and allows direct transformation from the received signals of the microphones into the driving signals of the loudspeakers. A model in which microphones are mounted on a rigid cylindrical baffle is introduced to stabilize the WFR filter. Numerical simulation results indicated that the reproduction accuracy in the neighboring region along the central axis of the cylindrical array was better preserved when using the proposed method than when the method with planar arrays was used.
Scientific journal, English - Real-Time Sound Field Transmission System by Using Wave Field Reconstruction Filter and Its Evaluation
Shoichi Koyama; Ken'ichi Furuya; Hisashi Uematsu; Yusuke Hiwasaki; Yoichi Haneda
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E97A, 9, 1840-1848, Sep. 2014, Peer-reviwed, A new real-time sound field transmission system is presented. To construct this system, a large listening area needs to be reproduced at not less than a constant height. Additionally, the driving signals of the loudspeakers should be obtained only from received signals of microphones. Wave field reconstruction (WFR) filtering for linear arrays of microphones and loudspeakers is considered to be suitable for this kind of system. An experimental system was developed to show the feasibility of real-time sound field transmission using the WFR filter. Experiments to measure the reproduced sound field and a subjective listening test of sound localization were conducted to evaluate the proposed system. Although the reproduced sound field included several artifacts such as spatial aliasing and faster amplitude decay, the experimental results indicated that the proposed system was able to provide sound localization accuracy for virtual sound sources comparable to that for real sound sources in a large listening area.
Scientific journal, English - Double-talk Robust Acoustic Echo Cancellation for CD-quality Hands-free Videoconferencing System
Masahiro Fukui; Suehiro Shimauchi; Yusuke Hioka; Akira Nakagawa; Yoichi Haneda
IEEE TRANSACTIONS ON CONSUMER ELECTRONICS, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 60, 3, 468-475, Aug. 2014, Peer-reviwed, A novel monaural acoustic echo cancellation method developed for a 20-kHz wideband hands-free video-or teleconferencing system is presented. The method can effectively reduce undesired acoustic echo included in a signal arriving at a microphone from a loudspeaker and can emphasize the target talker's voice when near-end and far-end talkers speak simultaneously (i.e. double-talk). The method estimates a power frequency response of an acoustic echo path between a loudspeaker and microphone (acoustic coupling level) whether or not double-talk has occurred, then calculates a post-filter that effectively reduces the undesired echo. In the echo cancellation processing, the calculation complexity is reduced to make the processing suitable for real-time implementation by using a low-complexity subband approach that employs a new subband filtering algorithm. Experiments were conducted to examine the performance of the proposed method. The results indicated that the proposed method delivered natural-sounding near-end speech even during double-talk periods while sufficiently suppressing the undesired echo(1).
Scientific journal, English - Adaptive Spectral Masking of AVQ Coding and Sparseness Detection for ITU-T G.711.1 Annex D and G.722 Annex B Standards
Masahiro Fukui; Shigeaki Sasaki; Yusuke Hiwasaki; Kimitaka Tsutsumi; Sachiko Kurihara; Hitoshi Ohmuro; Yoichi Haneda
IEICE TRANSACTIONS ON INFORMATION AND SYSTEMS, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E97D, 5, 1264-1272, May 2014, Peer-reviwed, We proposes a new adaptive spectral masking method of algebraic vector quantization (AVQ) for non-sparse signals in the modified discreet cosine transform (MDCT) domain. This paper also proposes switching the adaptive spectral masking on and off depending on whether or not the target signal is non-sparse. The switching decision is based on the results of MDCT-domain sparseness analysis. When the target signal is categorized as non-sparse, the masking level of the target MDCT coefficients is adaptively controlled using spectral envelope information. The performance of the proposed method, as a part of ITU-T G.711.1 Annex D, is evaluated in comparison with conventional AVQ. Subjective listening test results showed that the proposed method improves sound quality by more than 0.1 points on a five-point scale on average for speech, music, and mixed content, which indicates significant improvement.
Scientific journal, English - 球面調和関数展開に基づく2種類の超接話マイクロホンアレイ
羽田陽一; 古家賢一; 小山翔一; 丹羽健太
電子情報通信学会論文誌 A, 電子情報通信学会, J97-A, 4, 264-273, 01 Apr. 2014, Peer-reviwed
Scientific journal, Japanese - 雑音抑圧のための雑音対信号比率に基づく雑音パワー推定
福井勝宏; 島内末廣; 日岡裕輔; 中川 朗; 羽田陽一; 大室仲; 片岡章俊
Jounal of signal processing, 信号処理学会, 18, 1, 17-28, 25 Jan. 2014, Peer-reviwed
Scientific journal, Japanese - Real-Time Sound Field Transmission System by Using Wave Field Reconstruction Filter and Its Evaluation
KOYAMA Shoichi; FURUYA Ken'ichi; UEMATSU Hisashi; HIWASAKI Yusuke; HANEDA Yoichi
IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences, The Institute of Electronics, Information and Communication Engineers, 97, 9, 1840-1848, 2014, Peer-reviwed, A new real-time sound field transmission system is presented. To construct this system, a large listening area needs to be reproduced at not less than a constant height. Additionally, the driving signals of the loudspeakers should be obtained only from received signals of microphones. Wave field reconstruction (WFR) filtering for linear arrays of microphones and loudspeakers is considered to be suitable for this kind of system. An experimental system was developed to show the feasibility of real-time sound field transmission using the WFR filter. Experiments to measure the reproduced sound field and a subjective listening test of sound localization were conducted to evaluate the proposed system. Although the reproduced sound field included several artifacts such as spatial aliasing and faster amplitude decay, the experimental results indicated that the proposed system was able to provide sound localization accuracy for virtual sound sources comparable to that for real sound sources in a large listening area. - Sound field simulation for circular array based on spatial circular convolution
Yoichi Haneda; Ken'Ichi Furuya; Shoichi Koyama; Kenta Niwa; Kazunori Kobayashi
Acoustical Science and Technology, 日本音響学会, 35, 2, 99-107, 2014, Peer-reviwed, We propose a sound field simulation method for a circular array. In a linear array, when the distance between the loudspeaker units is equal to the interval between the observation points on the lines paralleled to the array, the sound pressures on the observation line can be calculated by the spatial convolution of the set of transfer functions and loudspeakers' driving signals. To apply this idea to a circular array, we developed a simulation method with equiangular observation points on the circle. The spatial circular convolution without zero padding, which is necessary in a linear array, can be used with this method. By conducting convolution based on the fast Fourier transform (FFT), the computational complexity is greatly reduced. Moreover, assuming that non-active loudspeakers are included in the loudspeaker array, the proposed method can be applied to an unequal interval array. For example, when the number of observation points is set to 128 and the number of loudspeakers is set to 32, circular convolution with FFT reduces the computational complexity to 75% compared to the conventional method. In addition, we argue that this method can be applied to a room in which the first reflected sounds are reflected from the floor. The proposed method is useful for simulating the sound field for a circular array when the suitable spatial samplings of the circumferential and radial directions are set. © 2014 The Acoustical Society of Japan.
Scientific journal, English - Wiener solution considering cross-spectral term between echo and near-end speech for acoustic echo reduction
Masahiro Fukui; Suehiro Shimauchi; Yusuke Hioka; Akira Nakagawa; Yoichi Haneda; Akitoshi Kataoka; Hitoshi Ohmuro
Acoustical Science and Technology, Acoustical Society of Japan, 35, 3, 150-158, 2014, Peer-reviwed, This paper introduces a frequency-domain acoustic echo reduction process based on a new Wiener-filtering method taking into account the cross-spectral term between the acoustic echo and the near-end speech. The conventional echo reduction method based on Wiener filtering estimates the gain based on the assumption that the cross-spectral term of the echo and the near-end speech is zero because the acoustic echo and the near-end speech are statistically uncorrelated. However, this assumption does not always hold true in practice because the gain is estimated in a very short period where the amount of statistical data, which is used to calculate the ensemble averages of the observed signals, is insufficient. As a result, the conventional method occasionally causes the perceptual degradation of sound quality during a double-talk situation
therefore, the performance is still not sufficient. Our goal was to accurately calculate the echo-reduction gain to decrease the speech distortions produced by the echo-reduction process. The proposed method solves a least mean square error of the Wiener-filtering method by taking into account the cross-spectral term between the echo and the near-end speech to obtain a better echo-reduction gain. The performance of this method was demonstrated by objective and subjective results in which speech distortions were decreased. ©2014 The Acoustical Society of Japan.
Scientific journal, English - CLOSE-TALKING SPHERICAL MICROPHONE ARRAY USING SOUND PRESSURE INTERPOLATION BASED ON SPHERICAL HARMONIC EXPANSION
Yoichi Haneda; Ken'ichi Furuya; Shoichi Koyama; Kenta Niwa
2014 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 604-608, 2014, Peer-reviwed, We propose a novel close-talking spherical microphone array that uses the residual signal between the observed sound pressure and the interpolated sound pressure at the center of the spherical array. The interpolated sound is obtained from the sound pressures observed on the surface of a sphere on the basis of the spherical harmonic expansion, assuming that the sound originates from the outside of the array. If the sound source is close to the spherical array, the array cannot express the spherical wave correctly because the number of microphones is limited. As a result, the residual signal increases. This method is a modified form of the conventional method, which interpolates the sound pressure by using the Kirchhoff integral equation. In contrast with the conventional method, we interpolate the sound at the center of the sphere by using only the average value of the sound pressures on the spherical array surface. The computer simulations were conducted using a 12-element spherical microphone array with radius of 5 cm. These results showed that the performances of both methods were almost equivalent, although the proposed method used half the number of microphones as the conventional method.
International conference proceedings, English - 音源の特性に応じた符号化モード選択多段量子化手法
福井 勝宏; 堤 公孝; 佐々木 茂明; 日和崎 祐介; 羽田 陽一
日本音響学会誌, 一般社団法人日本音響学会, 69, 12, 623-631, 01 Dec. 2013, Peer-reviwed, 高能率なベクトル量子化方式として,広く採用されているSVQ(Spherical Vector Quantization)やAVQ(Algebraic Vector Quantization)は,音声のスパース性に着目したコードブック設計により,音声の母音のようにスパースな表現を持つ入力信号に対し,少ない情報量で高品質に量子化できるという特徴を持つ。しかしながら,これらの方式では,楽音に含まれるブロードに広がったスペクトル成分など,入力信号がスパースでない場合の量子化においては,少ない情報量ですべての主要成分にエネルギーを割り当てることが難しく,復号信号にパワーの欠損が生じることがある。この問題を解決するため,本論文では,SVQやAVQに対し,入力信号がスパースでない場合に,ベクトル量子化でエネルギーが割り当てられなかった周波数成分を周波数包絡に基づく雑音成分で埋めることで,復号信号に生じるパワーの欠損を防ぐ量子化方式を提案する。主観評価試験から,様々な音源に対し,提案方式による量子化精度の向上を確認した。
Scientific journal, Japanese - 球調和関数展開に基づく多重極音源を用いた指向性合成
羽田陽一; 古家賢一; 島内末廣
日本音響学会誌, The Acoustical Society of Japan (ASJ), 69, 11, 577-588, Nov. 2013, Peer-reviwed, 音源の指向性を再現することや,任意の指向特性を合成することは,多くの応用において大変重要である。本論文では,球調和関数を用いて展開した指向特性を多重極の重ね合わせで表現する方法を提案し,球面スピーカアレー以外の形状でも球調和関数展開を用いた指向性合成が行えることを示す。また,2次までの球調和関数を表現するために,必要最低音源数である9個で構成した多重極音源を提案し,指向特性の合成シミュレーションを行った。その結果,音源間隔が波長に比べて小さい場合には少ない誤差で指向特性を合成可能であることを明らかにした。更に,スピーカ素子が軸上に並んだ直線アレーに対しても球調和関数展開に基づく多重極アレーを適用した。従来からある最小自乗法と指向特性を比較した結果,後方の指向特性は最小自乗法の方が良かったが,低周波数領域における側面の指向性は多重極アレーの方が良いことが分かった。これにより,直線アレーの設計においても球調和関数を用いた解析的な設計が有効であることが分かった。
Scientific journal, Japanese - Evaluation of microphone array based on diffused sensing with various filter design methods
K. Niwa; Y. Hioka; K. Kobayashi; K. Furuya; Y. Haneda
2013 Proceedings of the 21st European Signal Processing Conference (EUSIPCO), IEEE, 1-5, 09 Sep. 2013, Peer-reviwed
International conference proceedings, English - Real-Time Sound Field Transmission System by Using Wave Field Reconstruction Filter and Its Subjective Listening Test
S. Koyama; K. Furuya; H. Uematsu; Y. Hiwasaki; Y. Haneda
52nd International Conference: Sound Field Control - Engineering and Perception (September 2013), P-3, 02 Sep. 2013, Peer-reviwed
International conference proceedings, English - 音響エコー抑圧における結合量推定の追従性向上と高精度化
福井勝宏; 島内末廣; 日岡裕輔; 羽田陽一; 大室仲; 片岡彰俊
Journal of Signal Process., 17, 5, 167-177, Sep. 2013, Peer-reviwed
Scientific journal, Japanese - Underdetermined Sound Source Separation Using Power Spectrum Density Estimated by Combination of Directivity Gain
Y. Hioka; K. Furuya; K. Kobayashi; K. Niwa; Y. Haneda
IEEE Trans. Audio, Speech, and Language Processing, 21, 6, 1240 – 1250, Jun. 2013, Peer-reviwed
Scientific journal, English - Accurate Acoustic Echo Reduction with Residual Echo Power Estimation for Long Reverberation
M. Fukui; S. Shimauchi; Y. Hioka; H. Ohumuro; Y. Haneda
AES Convention:134th, 8891, 04 May 2013, Peer-reviwed
International conference proceedings, English - Analytical Approach to Wave Field Reconstruction Filtering in Spatio-Temporal Frequency Domain
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 21, 4, 685-696, Apr. 2013, Peer-reviwed, For transmission of a physical sound field in a large area, it is necessary to transform received signals of a microphone array into driving signals of a loudspeaker array to reproduce the sound field. We propose a method for transforming these signals by using planar or linear arrays of microphones and loudspeakers. A continuous transform equation is analytically derived based on the physical equation of wave propagation in the spatio-temporal frequency domain. By introducing spatial sampling, the uniquely determined transform filter, called a wave field reconstruction filter (WFR filter), is derived. Numerical simulations show that the WFR filter can achieve the same performance as that obtained using the conventional least squares (LS) method. However, since the proposed WFR filter is represented as a spatial convolution, it has many advantages in filter design, filter size, computational cost, and filter stability over the transform filter designed by the LS method.
Scientific journal, English - IMPROVEMENT USING CIRCULAR HARMONICS BEAMFORMING ON REVERBERATION PROBLEM OF WAVE FIELD RECONSTRUCTION FILTERING
Shoichi Koyama; Timothy Lee; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
2013 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 276-280, 2013, Peer-reviwed, In real-time sound field transmission systems, the driving signals of a loudspeaker array should be obtained using only the received signals of a microphone array. For efficient transformation of signals of planar or linear arrays, we previously proposed a method of applying a transform filter in the spatio-temporal frequency domain; this filter is defined as the wave field reconstruction (WFR) filter. In linear array configurations, the major artifact of this method is an increase in reverberation time and a decrease in the direct-to-reverberant energy ratio (DRR) because reflections from above and below the microphone array can not be distinguished. We propose a method combining circular harmonics beamforming with the WFR filter as a preprocess in order to match the reproduced DRR to the original one at the time the direct sound wave is properly reproduced. Simulation results indicated that the DRR reproduced using the proposed method was much closer to that reproduced by the method without beamforming.
International conference proceedings, English - SOUND FIELD REPRODUCTION USING MULTIPLE LINEAR ARRAYS BASED ON WAVE FIELD RECONSTRUCTION FILTERING IN HELICAL WAVE SPECTRUM DOMAIN
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda; Yoiti Suzuki
2013 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 271-275, 2013, Peer-reviwed, For with-height reproduction of a sound field, an efficient technique is to record and reproduce the sound field at lower resolution at an elevation angle than that at a horizontal angle based on auditory perception. To achieve this, developing a method using multiple horizontal linear arrays of microphones and loudspeakers is necessary. We propose a sound field reproduction method for a cylindrical array configuration which is based on a wave field reconstruction (WFR) filter analytically derived in the helical wave spectrum domain. The filter is stabilized by introducing a model in which microphones are mounted on a rigid cylindrical baffle. Numerical simulation results indicated that the reproduction accuracy of the proposed method was well preserved in the near-field region along the central axis of the cylinder even when the number of elements at angular position is small.
International conference proceedings, English - Sharp directive beamforming using microphone array and planar reflector
Kenta Niwa; Yusuke Hioka; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
Acoustical Science and Technology, 34, 4, 253-262, 2013, Peer-reviwed, A method for sharp directive beamforming using a reflector is proposed. The method involves physically varying the transfer functions and the spatial correlation matrix by placing a planar reflector near a linear microphone array. We confirmed through experiments that the proposed method was effective for increasing the spatial nulls and minimizing the power of interference in the beamforming output and consequently achieving sharp directivity. © 2013 The Acoustical Society of Japan.
Scientific journal, English - ACOUSTIC ECHO REDUCTION ROBUST AGAINST ECHO-PATH CHANGE WITH INSTANT ECHO-POWER-LEVEL ADJUSTMENT
Masahiro Fukui; Suehiro Shimauchi; Yusuke Hioka; Hitoshi Ohmuro; Yoichi Haneda
2013 PROCEEDINGS OF THE 21ST EUROPEAN SIGNAL PROCESSING CONFERENCE (EUSIPCO), IEEE, 1-5, 2013, Peer-reviwed, This paper proposes a new method of residual echo reduction (ER) to track abrupt increases in the residual echo. The method assumes that the spectral structure of the residual echo is maintained even if the residual echo abruptly increases; then, estimates only of the echo power level can be made in a short observation time. The proposed method instantaneously obtains the echo power level by focusing on all frequency-spectral components, and recalculates the echo component using bothe the obtained power level and the spectral structure estimated with the conventional method. The outstanding performance of the proposed method is demonstrated through objective and subjective experiments. The results revealed that the degradation in speech quality was mitigated even if the echo power level increased during the double-talk periods.
International conference proceedings, English - An estimation method of sound source orientation using eigenspace variation of spatial correlation matrix
Kenta Niwa; Yusuke Hioka; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences, Institute of Electronics, Information and Communication, Engineers, IEICE, E96-A, 9, 1831-1839, 2013, Peer-reviwed, A method to estimate sound source orientation in a reverberant room using a microphone array is proposed. We extend the conventional modeling of a room transfer function based on the image method in order to take into account the directivity of a sound source. With this extension, a transfer function between a sound source and a listener (or a microphone) is described by the superposition of transfer functions from each image source to the listener multiplied by the source directivity
thus, the sound source orientation can be estimated by analyzing how the image sources are distributed (power distribution of image sources) from observed signals. We applied eigenvalue analysis to the spatial correlation matrix of the microphone array observation to obtain the power distribution of image sources. Bsed on the assumption that the spatial correlation matrix for each set of source position and orientation is known a priori, the variation of the eigenspace can be modeled. By comparing the eigenspace of observed signals and that of pre-learned models, we estimated the sound source orientation. Through experiments using seven microphones, the sound source orientation was estimated with high accuracy by increasing the reverberation time of a room. Copyright © 2013 The Institute of Electronics, Information and Communication Engineers.
Scientific journal, English - MAP ESTIMATION OF DRIVING SIGNALS OF LOUDSPEAKERS FOR SOUND FIELD REPRODUCTION FROM PRESSURE MEASUREMENTS
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
2013 IEEE WORKSHOP ON APPLICATIONS OF SIGNAL PROCESSING TO AUDIO AND ACOUSTICS (WASPAA), IEEE, 1-4, 2013, Peer-reviwed, Sound field reproduction methods calculate driving signals of loudspeakers to reproduce the desired sound field. In common recording and reproduction systems, sound pressures at multiple positions obtained in a recording room are only known as the desired sound field; therefore, signal transformation algorithms from sound pressures into driving signals (SP-DS conversion) are necessary. Although several SP-DS conversion methods have been proposed, they do not take into account a priori information about the recorded sound field. However, approximate positions of sound sources can be obtained by using the received signals of microphones or other sensor data. We propose an SP-DS conversion method based on the maximum a posteriori (MAP) estimation when array configurations of the microphones and loudspeakers are planar or linear. The optimal basis functions and their coefficients for representing driving signals of the loudspeakers are optimized based on the prior information of the source positions. Numerical simulation results indicate that the proposed method can achieve higher reproduction accuracy compared to the current SP-DS conversion methods, especially in higher frequencies above the spatial Nyquist frequency.
International conference proceedings, English - Diffused sensing for sharp directive beamforming
Kenta Niwa; Yusuke Hioka; Kenichi Furuya; Yoichi Haneda
IEEE Transactions on Audio, Speech and Language Processing, IEEE, 21, 11, 2346-2355, 2013, Peer-reviwed, We generalized our previously proposed diffused sensing for a microphone array design to achieve sharp directive beamforming to enable various filter design methods to be applied. In the conventional microphone array, various filter design methods have been studied to narrow the directivity beam width. However, it is difficult to minimize the power of interference sources in the beamforming output (output interference power) over a broad frequency range since the cross-correlation between transfer functions from sound sources to microphones increases in some frequencies. With the diffused sensing, the cross-correlation is minimized by physically varying the transfer functions. We investigated how a microphone array should be designed in order to minimize the cross-correlation between transfer functions and found that placing the array in a diffuse acoustic field produces optimum results. Because the transfer functions are known a priori, this finding makes it possible to narrow the directivity beam width over a broad frequency range. This observation can be practically achieved by placing microphones inside a reflective enclosure, part of which is open to let sound waves enter. We conducted experiments using 24 microphones and confirmed that the output interference power was reduced over a broad frequency range and the beam width was narrowed by using the diffused sensing. © 2006-2012 IEEE.
Scientific journal, English - Sound field reproduction method in spatio-temporal frequency domain considering directivity of loudspeakers
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
132nd Audio Engineering Society Convention 2012, 885-892, 22 Oct. 2012, A method for transforming received signals of a microphone array into driving signals of a loudspeaker array for sound field reproduction is needed to achieve real-time sound field transmission systems from the far-end to the near-end. We recently proposed a transformmethod using planar or linear microphone and loudspeaker arrays in the spatio-temporal frequency domain, which is more efficient than conventional methods based on a least squares algorithm. In this method, the directivity of loudspeakers in the array is assumed to be omni-directional to derive the transform filter. However the directivity of common loudspeakers is not always omni-directional, especially at high frequencies. We therefore propose a transform method that takes into consideration the directivity of loudspeakers in the array, which is derived using analytical and numerical approaches. Numerical simulation results indicated that the accurately reproduced region of the proposed method was larger than that of the method with an omni-directional assumption. - Echo Reduction Using Wiener Gains Considering Short-Time Correlation Between Echo and Near-End Speech
M. Fukui; A. Nakagawa; S. Shimauchi; Y. Haneda; A. Kataoka
Proc. of IWAENC 2012, D-02, Sep. 2012, Peer-reviwed
International conference proceedings, English - Posterior Residual Echo Canceling and its Complexity Reduction in the Wave Domain
S. Emura; S. Koyama; K. Furuya; Y. Haneda
Proc. of IWAENC 2012, H-01, Sep. 2012, Peer-reviwed
International conference proceedings, English - Estimation of Direct-to-Reverberation Energy Ratio Based On Isotropic and Homogeneous Propagation Model
Y. Hioka; K. Furuya; K. Niwa; Y. Haneda
Proc. of IWAENC 2012, F-06, Sep. 2012
International conference proceedings, English - Nonlinear Acoustic Echo Cancellation Based on Piecewise Linear Approximation with Amplitude Threshold Decomposition
S. Shimauchi; Y. Haneda
Proc. of IWAENC 2012, B-02, Sep. 2012, Peer-reviwed
International conference proceedings, English - Superdirective Beamforming Using Microphone Array and Single Reflector
Kenta Niwa; Sumitaka Sakauchi; Ken'ichi Furuya; Manabu Okamoto; Yoichi Haneda
2012 IEEE INTERNATIONAL CONFERENCE ON CONSUMER ELECTRONICS (ICCE), IEEE, 17-18, 2012, Peer-reviwed, A method to generate superdirective beamforming using a microphone array and planar reflector is proposed. In this method, the beam width is reduced by positioning a reflector near the array and fixing the position of each microphone. Experimental results confirmed that the beam width was narrowed using the proposed method to half that with the conventional method.
International conference proceedings, English - Highly Realistic Collaboration System by using 4K Multi JPEG2000 Codecs and 6-Channel Acoustic Echo Canceller
SunYong Kim; Takayuki Nakachi; Tatsuya Fujii; Satoru Emura; Yoichi Haneda
IEEE INTERNATIONAL SYMPOSIUM ON INTELLIGENT SIGNAL PROCESSING AND COMMUNICATIONS SYSTEMS (ISPACS 2012), IEEE, 2012, Peer-reviwed, We develop a novel tele-collaboration system with low latency and high reliability by combining three 4K60p JPEG2000 video streaming systems and a 6-channel echo canceller. Our system realizes the synchronization of multiple video streams without even single frame delay. It can transmit 4K 60p streams with one-way latency of 80 msec, so users can communicate with one another with no unnatural pauses. Using a novel high-performance multi-channel acoustic echo canceller, our system can spatially localize each user's speech to the user's displayed position. We conduct a subjective assessment of our system through confrontational role-playing tasks and find that our system helps the subjects to share their feelings and atmosphere and enhances their cooperation.
International conference proceedings, English - DESIGN OF TRANSFORM FILTER FOR REPRODUCING ARBITRARILY SHIFTED SOUND FIELD USING PHASE-SHIFT OF SPATIO-TEMPORAL FREQUENCY
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
2012 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 381-384, 2012, Peer-reviwed, For real-time sound field transmission systems from a far-end to a near-end, the driving signals of a loudspeaker array at the near-end need to be calculated by using only received signals obtained by a microphone array at the far-end. Additionally, having the capability to control the location of the sound field to be reproduced in order to adjust it to the visual images is advantageous. The goal of this study was to develop a method to transform received signals of a microphone array into driving signals of a loudspeaker array in order to reproduce arbitrary shifted sound fields. We analytically derive a transform filter in the spatio-temporal frequency domain. The location of the sound field to be reproduced is controllable only by phase-shift of the transform filter. The proposed method was found to be computationally efficient compared to the conventional method based on a least squares algorithm, and numerical simulation results indicated that reproduction accuracies were almost the same in both methods.
International conference proceedings, English - LOCALIZED SOUND REPRODUCTION USING CIRCULAR LOUDSPEAKER ARRAY BASED ON ACOUSTIC EVANESCENT WAVE
Hiroaki Itou; Ken'ichi Furuya; Yoichi Haneda
2012 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 221-224, 2012, Peer-reviwed, We present a localized sound reproduction method in which sound energy is high in specified target areas and low everywhere else. Such a system needs to reproduce a sound which attenuates faster than a point source. Our study focuses on an acoustic evanescent wave that has faster distance attenuation. The evanescent wave can be generated when a wave transmitted at subsonic speed between two media. Previously, we used a linear loudspeaker array to reproduce the evanescent wave. Nevertheless, since this method requires a line source of infinite length, a large number of loudspeakers are needed to realize sufficiently fast distance attenuation. To reduce the number of loudspeakers, we propose an evanescent wave reproduction method by using a circular loudspeaker array. Computer simulation results show that our method achieves faster distance attenuation than a point source and reduces the number of loudspeakers to less than that of the linear loudspeaker array.
International conference proceedings, English - TELESCOPIC MICROPHONE ARRAY USING REFLECTOR FOR SEGREGATING TARGET SOURCE FROM NOISES IN SAME DIRECTION
Kenta Niwa; Yusuke Hioka; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
2012 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 5457-5460, 2012, Peer-reviwed, A spatial sensitivity control method for segregating the sound sources in the same direction by using an acoustic reflector is proposed. Our goal is to clearly pick up the target source at an arbitrary position using a microphone array. Though many methods have been studied for spatial sensitivity control, it is difficult to robustly suppress the power of noise sources in the same direction of the target source in a room. To overcome this problem, we attach a reflector to a microphone array to capture the reflected sounds whose characteristics vary depending on the distance from the array to the source. Assuming that the acoustical properties of the reflector are known e.g., measuring the transfer functions, those reflected sounds can be used as effective clues for segregating the sound sources in the same direction. With the proposed method, a filter for minimizing the output noise power is derived by taking into consideration of the acoustic properties of the reflector. Experiments were conducted in an actual room by using 96 microphones and a large reflector. We confirmed that the spatial sensitivities for segregating the target source at an arbitrary position from noise sources can be achieved by using the proposed method.
International conference proceedings, English - DIFFUSED SENSING FOR SHARP DIRECTIVITY MICROPHONE ARRAY
Kenta Niwa; Sumitaka Sakauchi; Ken'ichi Furuya; Manabu Okamoto; Yoichi Haneda
2012 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), IEEE, 225-228, 2012, Peer-reviwed, We propose a method for achieving sharp directivity by sensing signals in a diffuse acoustic field. Directivity control based on a beamforming method has been studied to make it possible to extract the waveform and location of an identified target source even if there are many noise sources. Sharp directivity can be achieved by minimizing the output noise power of a beamforming filter. However, it is difficult to minimize the output noise power over a broad frequency ranges. Our approach for minimizing the output noise power is to control the spatial properties of the transfer functions and the spatial correlation matrix, by using a reflector that surrounds a microphone array. We investigated the relationships between the output noise power and the structure of the spatial correlation matrix and found that it was possible to minimize the output noise power by sensing diffuse acoustic signals and by designing filters taking the diffuseness of the acoustic field into consideration. In experiments, we observed diffusely reflected signals by placing a truncated-octahedral reflector near a spherical microphone array. We designed filters by using measured transfer functions and confirmed that the proposed method was effective for reducing the output noise power and forming a sharp directivity beamforming filter.
International conference proceedings, English - Angular region-wise speech enhancement for hands-free speakerphone
Yusuke Hioka; Ken'ichi Furuya; Kazunori Kobayashi; Sumitaka Sakauchi; Yoichi Haneda
1st IEEE Global Conference on Consumer Electronics 2012, GCCE 2012, 58, 4, 144-145, 2012, Peer-reviwed, A novel speech enhancement technique using a microphone array is implemented on a hands-free speakerphone prototype. The technique is able to emphasise sound arriving from the angular region where speakers are located (speaker region) while effectively suppressing noise located in the other angular region (noise region). Experiments were performed to examine the performance of the proposed technique
the resultant directivity had a drastic drop-off between the regions while keeping its sensitivity constant within the same region. © 2012 IEEE.
International conference proceedings, English - Reproducing virtual sound sources in front of a loudspeaker array using inverse wave propagator
Shoichi Koyama; Ken'Ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
IEEE Transactions on Audio, Speech and Language Processing, 20, 6, 1746-1758, 2012, Peer-reviwed, It has been possible to reproduce point sound sources between listeners and a loudspeaker array by using the focused-source method. However, this method requires physical parameters of the sound sources to be reproduced, such as source positions, directions, and original signals. This fact makes it difficult to apply the method to real-time reproduction systems because decomposing received signals into such parameters is not a trivial task. This paper proposes a method for recreating virtual sound sources in front of a planar or linear loudspeaker array. The method is based on wave field synthesis but extended to include the inverse wave propagator often used in acoustical holography. Virtual sound sources can be placed between listeners and a loudspeaker array even when the received signals of a microphone array equally aligned with the loudspeaker array are only known. Numerical simulation results are presented to compare the proposed and focused-source methods. A system was constructed consisting of linear microphone and loudspeaker arrays and measurement experiments were conducted in an anechoic room. When comparing the sound field reproduced using the proposed method with that using the focused-source method, it was found that the proposed method could reproduce the sound field at almost the same accuracy. © 2006-2012 IEEE.
Scientific journal, English - Inverse wave propagation for reproducing virtual sources in front of loudspeaker array
Shoichi Koyama; Yusuke Hiwasaki; Ken'ichi Furuya; Yoichi Haneda
European Signal Processing Conference, 1322-1326, 01 Dec. 2011, This paper presents a new combination of wave field synthesis with inverse wave propagation that can recreate virtual sound sources in front of a loudspeaker array. Wave field synthesis is a well-known sound field reproduction technique, based on the Rayleigh integral, that reconstructs sound pressure distribution by using a planar or linear loudspeaker array. We show that, using a holographic approach, the re-construction position can be displaced towards the listener, in front of the secondary sources. As a result, virtual primary sources can be placed between the listener and the secondary sources. Numerical simulation results are presented to show the efficacy of the proposed method. We implemented an experimental system using linear microphone and loudspeaker arrays to reproduce the sound field in a real environment. Results of perceptual experiments showed that the proposed method can achieve sound localization accuracy for virtual sound sources equivalent to that for real sound sources. © 2011 EURASIP. - Estimating Direct-to-Reverberant Energy Ratio Using D/R Spatial Correlation Matrix Model
Yusuke Hioka; Kenta Niwa; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 19, 8, 2374-2384, Nov. 2011, Peer-reviwed, We present a method for estimating the direct-to-reverberant energy ratio (DRR) that uses a direct and reverberant sound spatial correlation matrix model (Hereafter referred to as the spatial correlation model). This model expresses the spatial correlation matrix of an array input signal as two spatial correlation matrices, one for direct sound and one for reverberation. The direct sound propagates from the direction of the sound source but the reverberation arrives from every direction uniformly. The DRR is calculated from the power spectra of the direct sound and reverberation that are estimated from the spatial correlation matrix of the measured signal using the spatial correlation model. The results of experiment and simulation confirm that the proposed method gives mostly correct DRR estimates unless the sound source is far from the microphone array, in which circumstance the direct sound picked up by the microphone array is very small. The method was also evaluated using various scales in simulated and actual acoustical environments, and its limitations revealed. We estimated the sound source distance using a small microphone array, which is an example of application of the proposed DRR estimation method.
Scientific journal, English - Sound Field Recording and Reproduction Using Transform Filter Designed in Spatio-Temporal Frequency Domain
S. Koyama; K. Furuya; Y. Hiwasaki; Y. Haneda
AES the 131st Convention, P-19-2, Oct. 2011, Peer-reviwed
International conference proceedings, English - Measuring Sweeping Echoes in Rectangular Cross-Section Reverberant Fields
Kenji Kiyohara; Ken'ichi Furuya; Yoichi Haneda; Yutaka Kaneda
ACTA ACUSTICA UNITED WITH ACUSTICA, S HIRZEL VERLAG, 97, 2, 278-283, Mar. 2011, Peer-reviwed, We investigated a new acoustical phenomenon, which we call sweeping echoes, in a two-dimensional (2D) space. Sweeping echoes in a three-dimensional (3D) space have recently been reported. We first investigated the regularity of reflected sound in a 2D regularly shaped space based on number theory. The reflected pulse sound train has almost equal intervals between pulses on the squared-time axis as in a 3D space. This regularity of the arrival time of reflected pulse sounds generates sweeping echoes whose frequencies increase linearly with time. Computer simulation of room acoustics shows good agreement with the theoretical results. We first describe our investigation of a square cross-section based on number theory. Next, we describe rectangular cross-sections with various aspect ratios investigated based on the same theory as that for the square. We also discuss our measurements of sweeping echoes in a long hallway. We propose a method for extracting sweep rates of sweeping echoes by calculating their correlation with a time stretched pulse. We analyzed the sweeping echoes for a source and receiver at the center of a rectangular cross-section. These sweeping echoes were not only perceived at the exact center position but also around the center.
Scientific journal, English - A METHOD FOR POSTERIOR FREQUENCY-DOMAIN MULTI-CHANNEL RESIDUAL ECHO CANCELING
Satoru Emura; Yoichi Haneda
2011 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, IEEE, 421-424, 2011, Peer-reviwed, We propose a posterior frequency-domain multi-channel residual echo canceling method. As the number of reproduction channels of an echo canceller increases, its adaptive filter shows a slower convergence of the filter coefficient error, that is, large echo path mismatch. In this situation, the change in the human talker in a remote site produces an abrupt residual echo more frequently. The proposed method immediately cancels this abrupt residual echo by estimating it from loudspeaker-reproduced signals, adaptive filter output, and echo replica with a much shorter frame length compared with adaptive filter length. By taking the echo replica into account, this frequency-domain method can improve residual-echo estimates. Another feature of the proposed method is controlling the residual echo estimates according to the confidence interval of these estimates.
International conference proceedings, English - DESIGN OF MULTIPOLE LOUDSPEAKER ARRAY BASED ON SPHERICAL HARMONIC EXPANSION
Yoichi Haneda; Ken'ichi Furuya; Hiroaki Itou
2011 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, IEEE, 141-144, 2011, Peer-reviwed, We investigated the multipole loudspeaker array based on spherical harmonic expansion in the Cartesian coordinate system. The arrangement of the multipoles with the least number of loudspeaker units has been studied for 2nd order of spherical harmonic expansion. The directivity of the multipole array does not achieve ideal directivity of the spherical functions due to the asymmetrical arrangement. The multipole array was also applied to an end-fire array because a multipole array uses less loudspeakers than the spherical array if the look-direction is fixed. We compared the performance of the conventional least-squares design method and multipole array method for an end-fire array. The least squares method had a higher directivity due to the large number of variable coefficients. Nevertheless, the multipole array based on spherical harmonic expansion has advantages of easy setup in the Cartesian coordinate system, analytical directivity, and small number of filter coefficients.
International conference proceedings, English - Dual-mode AVQ Coding Based on Spectral Masking and Sparseness Detection for ITU-T G.711.1/G.722 Super-wideband Extensions
Masahiro Fukui; Shigeaki Sasaki; Yusuke Hiwasaki; Kurihara Sachiko; Yoichi Haneda
12TH ANNUAL CONFERENCE OF THE INTERNATIONAL SPEECH COMMUNICATION ASSOCIATION 2011 (INTERSPEECH 2011), VOLS 1-5, ISCA-INT SPEECH COMMUNICATION ASSOC, 2536-2539, 2011, Peer-reviwed, ITU-T Recommendations G.711.1 Annex D and G.722 Annex B, which are super-wideband (50-14,000 Hz) extensions to G.711.1 and G.722, have been recently standardized. This paper introduces a new coding method proposed and employed in the above ITU-T standards. The proposed coding method employs an adaptive spectral masking of the algebraic vector quantization (AVQ) for MDCT-domain non-sparse signals. The adaptive spectral masking is switched on and off based on MDCT-domain sparseness analysis. When the target MDCT coefficients are categorized as non-sparse, masking level of the target MDCT coefficients is adaptively controlled using spectral envelope information. The performance of the proposed method as a part of the ITU-T G.711.1 Annex D is evaluated in comparison with the ordinary AVQ. Subjective listening test results show that the proposed method improves the sound quality more than 0.1 points with a five grade scale in average of speech, music and mixed content, and the significance of the improvement is validated.
International conference proceedings, English - INVERSE WAVE PROPAGATION FOR REPRODUCING VIRTUAL SOURCES IN FRONT OF LOUDSPEAKER ARRAY
Shoichi Koyama; Yusuke Hiwasaki; Ken'ichi Furuya; Yoichi Haneda
19TH EUROPEAN SIGNAL PROCESSING CONFERENCE (EUSIPCO-2011), EUROPEAN ASSOC SIGNAL SPEECH & IMAGE PROCESSING-EURASIP, 1322-1326, 2011, Peer-reviwed, This paper presents a new combination of wave field synthesis with inverse wave propagation that can recreate virtual sound sources in front of a loudspeaker array. Wave field synthesis is a well-known sound field reproduction technique, based on the Rayleigh integral, that reconstructs sound pressure distribution by using a planar or linear loudspeaker array. We show that, using a holographic approach, the reconstruction position can be displaced towards the listener, in front of the secondary sources. As a result, virtual primary sources can be placed between the listener and the secondary sources. Numerical simulation results are presented to show the efficacy of the proposed method. We implemented an experimental system using linear microphone and loudspeaker arrays to reproduce the sound field in a real environment. Results of perceptual experiments showed that the proposed method can achieve sound localization accuracy for virtual sound sources equivalent to that for real sound sources.
International conference proceedings, English - EVANESCENT WAVE REPRODUCTION USING LINEAR ARRAY OF LOUDSPEAKERS
Hiroaki Itou; Ken'ichi Furuya; Yoichi Haneda
2011 IEEE WORKSHOP ON APPLICATIONS OF SIGNAL PROCESSING TO AUDIO AND ACOUSTICS (WASPAA), IEEE, 37-40, 2011, Peer-reviwed, An acoustic evanescent wave reproduction method is presented. Evanescent waves have faster distance attenuation. In most sound field reproduction approaches, the evanescent wave is a spatial aliasing component and is an undesirable wave field for reproducing a pure plane wave. However, because of its faster distance attenuation property, it does not affect these approaches. In this work, we attempt to reproduce only the evanescent wave. If only the evanescent wave is reproduced, it can be applied for a local area sound reproduction system that can localize a sound field around one specific location. The evanescent wave reproduction system is constructed by using a linear loudspeaker array and digital filters. Computer simulation results showed that our method achieves faster distance attenuation than a simple sound source.
International conference proceedings, English - DESIGN OF TRANSFORM FILTER FOR SOUND FIELD REPRODUCTION USING MICROPHONE ARRAY AND LOUDSPEAKER ARRAY
Shoichi Koyama; Ken'ichi Furuya; Yusuke Hiwasaki; Yoichi Haneda
2011 IEEE WORKSHOP ON APPLICATIONS OF SIGNAL PROCESSING TO AUDIO AND ACOUSTICS (WASPAA), IEEE, 5-8, 2011, Peer-reviwed, In this paper, we propose a novel method of sound field reproduction using a microphone array and loudspeaker array. Our objective is to obtain the driving signal of a planar or linear loudspeaker array only from the sound pressure distribution acquired by the planar or linear microphone array. In this study, we derive a formulation of the transform from the received signals of the microphone array to the driving signals of the loudspeaker array. The transform is achieved as a mean of a filter in a spatio-temporal frequency domain. Numerical simulation results are presented to compare the proposed method with the method based on the conventional least square algorithm. The reproduction accuracies were found to be almost the same, however, the filter size and amount of calculation required for the proposed method were much smaller than those for the least square algorithm based one.
International conference proceedings, English - Evaluating small end-fire loudspeaker array under various reverberations
Yoichi Haneda; Ken'ichi Furuya; Akitoshi Kataoka
Acoustical Science and Technology, 32, 6, 244-250, 2011, Peer-reviwed, A compact end-fire loudspeaker array based on the linearly constrained minimum variance has been investigated in various reverberation rooms. In the linearly constrained minimum variance beamformer, the coherent matrix that determines the suppression direction of the radiation sound is important. Therefore, we evaluated the directivities of a loudspeaker array for four different coherent matrices that were calculated using the following methods in which the array will not radiate the sound: (I) for any direction except the reproduction direction, (II) in the direction of several measured room transfer functions, (III) in the direction of several ideal (free field) room transfer functions, and (IV) in the direction of the weighted ideal room transfer functions. To evaluate performance, a 4-element line array of 28-mm-diameter loudspeakers with equal spacing of 48mm was implemented. The array used loudspeaker units without enclosures to shorten the spaces between the units. The directivities of the prototype loudspeaker array were measured in an anechoic room and three reverberant rooms by using the above four methods. The prototype arrays achieved higher directivity of 6 to 15 dB for wideband frequencies (300 Hz to 3.4 kHz) than the single loudspeaker for all methods. Moreover, the experimental results clarified that the array with method (IV), which did not use prior measured room transfer functions, had efficient performance in comparison with the array with method (II), especially when the reverberation time was 120 ms or less. © 2011 The Acoustical Society of Japan.
Scientific journal, English - Improving Power Spectra Estimation in 2-Dimensional Areas Using Number of Active Sound Sources
Yusuke Hioka; Ken'ichi Furuya; Yoichi Haneda; Akitoshi Kataoka
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E94A, 1, 273-281, Jan. 2011, Peer-reviwed, An improvement of estimating sound power spectra located in a particular 2-dimensional area is proposed. We previously proposed a conventional method that estimates sound power spectra using multiple fixed beamformings in order to emphasize speech located in a particular 2-dimensional area. However, the method has one drawback that the number of areas where the active sound sources are located must be restricted. This restriction makes the method less effective when many noise source located in different areas are simultaneously active. In this paper, we reveal the cause of this restriction and determine the maximum number of areas for which the method is able to simultaneously estimate sound power spectra. Then we also introduce a procedure for investigating areas that include active sound sources to reduce the number of unknown power spectra to be estimated. The effectiveness of the proposed method is examined by experimental evaluation applied to sounds recorded in a practical environment.
Scientific journal, English - Evaluating estimation of direct-to-reverberation energy ratio using D/R spatial correlation matrix model
Yusuke Hioka; Kenta Niwa; Sumitaka Sakauchi; Ken' ichi Furuya; Yoichi Haneda
20th International Congress on Acoustics 2010, ICA 2010 - Incorporating Proceedings of the 2010 Annual Conference of the Australian Acoustical Society, 4, 2745-2751, 01 Dec. 2010, We evaluate the accuracy of direct-to-reverberation energy ratio (DRR) estimation that uses the direct sound to reverberation spatial correlation matrix model (DRSC model). The DRSC model, which expresses the spatial correlation matrix with two different matrices of direct sound and reverberation, assumes that the direct sound propagates only from the direction of the sound source but that the reverberation arrives from every direction uniformly. The DRR is calculated from the power spectra of both the direct sound and reverberation that are estimated from the spatial correlation matrix of the observed signal. The method was evaluated using various scales in both simulated and actual acoustical environments. The evaluation results confirmed the effectiveness of DRR estimation using the DRSC model and also revealed its limitations. Copyright © (2010) by the International Congress on Acoustics. - Evaluating Estimation of Direct-to-Reverberation Energy Ratio using D/R Spatial Correlation Matrix Model
Y. Hioka; K. Niwa; S. Sakauchi; K. Furuya; Y. Haneda
International Commission for Acoustics (ICA), 950-956, Aug. 2010, Peer-reviwed
International conference proceedings, English - ESTIMATION OF SOUND SOURCE ORIENTATION USING EIGENSPACE OF SPATIAL CORRELATION MATRIX
Kenta Niwa; Yusuke Hioka; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
2010 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, IEEE, 129-132, 2010, Peer-reviwed, We propose a method for estimating the sound source orientation by using the reflection sounds. The sound source orientation is important spatial information for promoting communication using teleconference systems. We assume that the observed signals captured using several microphones in a reverberant room are used for estimating the sound source orientation. Since the power of each reflection sound depends on the sound source orientation, the transfer functions between a sound source and multiple microphones are varied corresponding to the sound source orientation. We found that the eigenspace of spatial correlation constructed from the observed signals has a characteristic shape corresponding to the sound source orientation. We also proposed an efficient method for estimating the sound source orientation by matching the eigenspace of observed signals with pre-learned eigenspace models for every sound source orientation. In numerical experiments, we obtained about 80% accuracy. We confirmed the effectiveness of the proposed method.
International conference proceedings, English - ESTIMATING DIRECT-TO-REVERBERANT ENERGY RATIO BASED ON SPATIAL CORRELATION MODEL SEGREGATING DIRECT SOUND AND REVERBERATION
Yusuke Hioka; Kenta Niwa; Sumitaka Sakauchi; Ken'ichi Furuya; Yoichi Haneda
2010 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, IEEE, 149-152, 2010, Peer-reviwed, A new approach for estimating the direct-to-reverberant energy ratio (DRR) using a microphone array is proposed. The method is based on amodel of a spatial correlation matrix that segregates direct sound and reverberation. It estimates DRR from the power spectra of both components, which are derived from the correlation matrix of the observed signal. In experiments performed in simulated and actual reverberant environments, the proposed method mostly succeeded in estimating DRR accurately. We also present speech enhancement using binary masking as an example of an application of the estimated DRR. By utilization of the DRR as a factor to discriminate the distances of speakers, separation of speech signals whose sources were located in the same direction but at different distances was achieved.
International conference proceedings, English - 重み付き最小二乗逆フィルタ計算による雑音・残響環境下での音声強調処理の品質改善
古家賢一; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J93-A, 6, 387-396, 2010, Peer-reviwed
Scientific journal, Japanese - Inverse Wave Propagation in Wave Field Synthesis
S. Koyama; K. Furuya; Y. Haneda; Y. Hiwasaki
Audio Engineering Society (AES) International Conference, P-10, 2010, Peer-reviwed
International conference proceedings, English - 20-kHz Frequency-Range Acoustic Echo Canceller For High-Quality TV Conferencing
Masahiro Fukui; Akira Nakagawa; Suehiro Shimauchi; Yoichi Haneda; Akitoshi Kataoka
2009 IEEE INTERNATIONAL CONFERENCE ON CONSUMER ELECTRONICS, IEEE, -, 213-+, 2009, Peer-reviwed, This paper presents an overview of a new sub-band acoustic echo canceller for the 100 Hz-20 kHz frequency band that we have developed. It consists mainly of adaptive digital filter and echo reduction processes to eliminate an echo signal. The experimental results indicate that the new echo canceller suppresses the echo signal at least over 35 dB and delivers natural near-end speech sound even during double-talk situations.
International conference proceedings, English - New Acoustic Echo Canceller for Videotelephony-Enabled Wideband Business Phone
Masahiro Fukui; Akira Nakagawa; Suehiro Shimauchi; Yoichi Haneda; Takanori Somei; Hiroyasu Fuyuki; Akitoshi Kataoka
ISCE: 2009 IEEE 13TH INTERNATIONAL SYMPOSIUM ON CONSUMER ELECTRONICS, VOLS 1 AND 2, IEEE, 372-+, 2009, Peer-reviwed, The implementation and evaluation of our new acoustic echo canceller for a videotelephony-enabled business phone is presented. The developed echo canceller has the following features, the sufficient echo and noise suppression, low processing delay, natural-sounding processed speech and low computational cost. The experimental results demonstrate that the new echo canceller suppresses the echo and noise sufficiently and thus achieves natural-sounding speech.
International conference proceedings, English - Speech enhancement in a 2-dimensional area based on power spectrum estimation of multiple areas with investigation of existence of active sources
Yusuke Hioka; Ken'ichi Furuya; Yoichi Haneda; Akitoshi Kataoka
INTERSPEECH 2009: 10TH ANNUAL CONFERENCE OF THE INTERNATIONAL SPEECH COMMUNICATION ASSOCIATION 2009, VOLS 1-5, ISCA-INT SPEECH COMMUNICATION ASSOC, 1331-+, 2009, Peer-reviwed, A microphone array that emphasizes sound sources located in a particular 2-dimensional area is described. We previously developed a method that estimates the power spectra of target and noise sounds using multiple fixed beamformings. However, that method requires the areas where the noise sources are located to be restricted. We describe the principle of this limitation then propose a procedure that investigates the possibility of the existence of a sound source in a target area and other areas beforehand to reduce the number of unknown power spectra to be estimated.
International conference proceedings, English - Dynamic impulse response model for nonlinear acoustic system and its application to acoustic echo canceller
Shoichiro Saito; Akira Nakagawa; Yoichi Haneda
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, -, 201-204, 2009, Peer-reviwed, We proposed a dynamic impulse response model for a nonlinear acoustic system and a nonlinear acoustic echo canceller based on this model. In an acoustic system with small loudspeakers and low-quality enclosures, acoustic signal paths between loudspeakers and microphones often include nonlinearity. Such paths cannot be represented with a conventional linear system. We investigated the impulse responses with various input signal levels and find that the responses differed with regularity in accordance with the level of the input signal. Thus, we propose a dynamic impulse response model based on this fact. In this model, the system response varies in accordance with the input signal levels and the nonlinearity can be approximated with a set of linear filters. We applied this model to an acoustic echo canceller, and experimental evaluations using a small acoustic system show that the echo cancellation performance of the proposed method is improved relative to that of the conventional method by 4 dB on average and 10 dB at maximum. ©2009 IEEE.
International conference proceedings, English - Robust frequency domain acoustic echo cancellation filter employing normalized residual echo enhancement
Suehiro Shimauchi; Yoichi Haneda; Akitoshi Kataoka
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E91A, 6, 1347-1356, Jun. 2008, Peer-reviwed, We propose a new robust frequency domain acoustic echo cancellation filter that employs a normalized residual echo enhancement. By interpreting the conventional robust step-size control approaches as a statistical-model-based residual echo enhancement problem, the optimal step-size introduced in the most of conventional approaches is regarded as optimal only on the assumption that both the residual echo and the outlier in the error output signal are described by Gaussian distributions. However, the Gaussian-Gaussian mixture assumption does not always hold well, especially when both the residual echo and the outlier are speech signals (known as a double-talk situation). The proposed filtering scheme is based on the Gaussian-Laplacian mixture assumption for the signals normalized by the reference input signal amplitude. By comparing the performances of the proposed and conventional approaches through the simulations, we show that the Gaussian-Laplacian mixture assumption for the normalized signals can provide a better control scheme for the acoustic echo cancellation.
Scientific journal, English - A display-mounted high-quality stereo microphone array for high-definition videophone system
Yusuke Hioka; Manabu Okamoto; Kazunori Kobayashi; Yoichi Haneda; Akitoshi Kataoka
IEEE TRANSACTIONS ON CONSUMER ELECTRONICS, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 54, 2, 778-786, May 2008, Peer-reviwed, A stereo microphone array developed for a high-definition videophone system is presented. The array consists of a pair of fixed beamformers to collect sounds in stereo clearly while suppressing the far-end talk emitted from the loudspeakers to decrease the undesirable influence of acoustic echo. Both the objective and subjective experimental results demonstrate that the microphone array satisfies the required specifications, which have not been achieved by several conventional schemes.
Scientific journal, English - A hands-free unit with noise reduction by using adaptive beamformer
Kazunori Kobayashi; Yoichi Haneda; Ken'ichi Furuya; Akitoshi Kataoka
IEEE TRANSACTIONS ON CONSUMER ELECTRONICS, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 54, 1, 116-122, Feb. 2008, Peer-reviwed, This paper presents an implementation of an adaptive beamformer in a hands-free unit. The proposed adaptive beamformer suppresses stationary noise and interference sound. The adaptive beamformer and acoustic echo canceller were implemented in a compact hands-free unit with a low-cost DSP. Experimental results demonstrate the noise reduction performance of the hands-free unit(1).
Scientific journal, English - Noise robust speech dereverberation using constrained inverse filter
Ken'ichi Furuya; Akitoshi Kataoka; Yoichi Haneda
INTERSPEECH 2008: 9TH ANNUAL CONFERENCE OF THE INTERNATIONAL SPEECH COMMUNICATION ASSOCIATION 2008, VOLS 1-5, ISCA-INT SPEECH COMMUNICATION ASSOC, 427-+, 2008, Peer-reviwed, A noise robust dereverberation method is presented for speech enhancement in noisy reverberant conditions. This method introduces the constraint of minimizing the noise power in the inverse filter computation of dereverberation. It is shown that there exists a tradeoff between reducing the reverberation and reducing the noise; this tradeoff can be controlled by the constraint. Inverse filtering reduces early reflections and directional noise. In addition, spectral subtraction is used to suppress the tail of the inverse-filtered reverberation and residual noise. The performance of our method is objectively and subjectively evaluated in experiments using measured room impulse responses. The results indicate that this method provides better speech quality than the conventional methods.
International conference proceedings, English - A hands-free unit with noise reduction by using adaptive beamformer
Kazunori Kobayashi; Yoichi Haneda; Ken'ichi Furuya; Akitoshi Kataoka
2008 DIGEST OF TECHNICAL PAPERS INTERNATIONAL CONFERENCE ON CONSUMER ELECTRONICS, IEEE, -, 35-36, 2008, Peer-reviwed, This paper presents an implementation of adaptive beamformer in a hands-free unit. The proposed adaptive beamformer suppresses stationary noise and interference sound. The adaptive beamformer and acoustic echo canceller were implemented in a compact hands-free unit with a low-cost DSP. The experimental results demonstrate the noise reduction performance of the hands-free unit.
International conference proceedings, English - A display-mounted high-quality stereo microphone array for high-definition videophone system
Yusuke Hioka; Manabu Okamoto; Kazunori Kobayashi; Yoichi Haneda; Akitoshi Kataoka
2008 DIGEST OF TECHNICAL PAPERS INTERNATIONAL CONFERENCE ON CONSUMER ELECTRONICS, IEEE, -, 21-22, 2008, Peer-reviwed, A stereo microphone array developed for a high-definition videophone system is presented. It consists of a pair of fixed beamformers to collect sounds in stereo clearly while suppressing the far-end talk to decrease the undesirable influence of acoustic echo. Experimental results prove that the array satisfies the required specifications, which have not been achieved with the conventional schemes.
International conference proceedings, English - A display-mounted high-quality stereo microphone array for high-definition videophone system
Yusuke Hioka; Manabu Okamoto; Kazunori Kobayashi; Yoichi Haneda; Akitoshi Kataoka
2008 DIGEST OF TECHNICAL PAPERS INTERNATIONAL CONFERENCE ON CONSUMER ELECTRONICS, IEEE, 54, 2, 21-22, 2008, Peer-reviwed, A stereo microphone array developed for a high-definition videophone system is presented. It consists of a pair of fixed beamformers to collect sounds in stereo clearly while suppressing the far-end talk to decrease the undesirable influence of acoustic echo. Experimental results prove that the array satisfies the required specifications, which have not been achieved with the conventional schemes.
International conference proceedings, English - 位置が未知である複数マイクロホンアレーと距離測定スピーカを用いた音源位置推定
小林和則; 古家賢一; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J91-A, 11, 1006-1016, 2008, Peer-reviwed
Scientific journal, Japanese - An approach to solve local minimum problem in sound source and microphone localization
Kazunori Kobayashi; Ken'ichi Furuya; Yoichi Haneda; Akitoshi Kataoka
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E90A, 12, 2826-2834, Dec. 2007, Peer-reviwed, We previously proposed a method of sound source and microphone localization. The method estimates the locations of sound sources and microphones from only time differences of arrival between signals picked up by microphones even if all their locations are unknown. However, there is a problem that some estimation results converge to local minimum solutions because this method estimates locations iteratively and the error function has multiple minima. In this paper, we present a new iterative method to solve the local minimum problem. This method achieves accurate estimation by selecting effective initial locations from many random initial locations. The computer simulation and experimental results demonstrate that the presented method eliminates most local minimum solutions. Furthermore, the computational complexity of the presented method is similar to that of the previous method.
Scientific journal, English - Gradient-limited affine projection algorithm for double-talk-robust and fast-converging acoustic echo cancellation
Suehiro Shimauchi; Yoichi Haneda; Akitoshi Kataoka; Akinori Nishihara
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E90A, 3, 633-641, Mar. 2007, Peer-reviwed, We propose a gradient-limited affine projection algorithm (GL-APA), which can achieve fast and double-talk-robust convergence in acoustic echo cancellation. GL-APA is derived from the M-estimationbased nonlinear cost function extended for evaluating multiple error signals dealt with in the affine projection algorithm (APA). By considering the non-linearity of the gradient, we carefully formulate an update equation consistent with multiple input-output relationships, which the conventional APA inherently satisfies to achieve fast convergence. We also newly introduce a scaling rule for the nonlinearity, so we can easily implement GL-APA by using a predetermined primary function as a basis of scaling with any projection order. This guarantees a linkage between GL-APA and the gradientlimited normalized least-mean-squares algorithm (OL-NLMS), which is a conventional algorithm that corresponds to the GL-APA of the first order. The performance of GL-APA is demonstrated with simulation results.
Scientific journal, English - New stereo echo canceller operating on single digital signal processor
Satoru Emura; Suehiro Shimauchi; Sumitaka Sakauchi; Youichi Haneda; Junji Kojima; Akitoshi Kataoka
Acoustical Science and Technology, 28, 3, 172-180, 2007, Peer-reviwed, In this paper, we provide an overview of a new immersive, cost-effective stereo echo canceller that we have developed recently. To achieve immersive stereo hands-free communication, we expanded the frequency range from 0.1-7 kHz to 0.1-20 kHz and revised the echo reduction processing. To achieve a cost-effective canceller, we revised the adaptive algorithm to reduce the required memory and implemented the entire signal processing in a single fixed-point digital signal processor (DSP). The experiments indicate that the new stereo echo canceller delivers near-end speech and background sound more naturally under the double-talk situation. © 2007 The Acoustical Society of Japan.
Scientific journal, English - ステレオエコーキャンセラのためのチャネル間相関変動を強調する周波数領域適応アルゴリズム
江村暁; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J90-A, 6, 506-516, 2007, Peer-reviwed
Scientific journal, Japanese - Stereo echo cancellation algorithm using adaptive update on the basis of enhanced input-signal vector
S Emura; Y Haneda; A Kataoka; S Makino
SIGNAL PROCESSING, ELSEVIER SCIENCE BV, 86, 6, 1157-1167, Jun. 2006, Peer-reviwed, Stereo echo cancellation requires a fast converging adaptive algorithm because the stereo input signals are highly cross correlated and the convergence rate of the misalignment is slow even after preprocessing for unique identification of stereo echo paths. To speed up the convergence, we propose enhancing the contribution of the decorrelated components in the preprocessed input-signal vector to adaptive updates. The adaptive filter coefficients are updated on the basis of either a single or multiple past enhanced input-signal vectors.
For a single-vector update, we show how this enhancement improves the convergence rate by analyzing the behavior of the filter coefficient error in the mean. For a two-past-vector update, simulation showed that the proposed enhancement leads to a faster decrease in misalignment than the corresponding conventional second-order affine projection algorithm while computational complexities are almost the same. (c) 2005 Elsevier B.V. All rights reserved.
Scientific journal, English - Noise reduction with less processing delay for analysis and synthesis using a single microphone
S Sakauchi; Y Haneda; A Kataoka
ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE, SCRIPTA TECHNICA-JOHN WILEY & SONS, 89, 2, 24-32, 2006, Peer-reviwed, To realize comfortable voice communication with reduced steady ambient noise such as air conditioning noise, there is a form of noise suppression processing based on short-time spectral amplitude estimation. This is a frequency domain process using one microphone. Since the amount of processing is rather small, this is an extremely useful method for realization of equipment and applications. However, the processing delay associated with frame processing in the frequency domain causes deterioration of the voice communications quality. To deal with this problem, the present research proposes a scheme to reduce the processing delay. In the proposed system, deterioration of the processing quality is prevented by retaining the frequency resolution and the processing delay is reduced by compressing the shift width of the processing frame by performing overlap-addition. Objective evaluation confirms that the processing delay can be reduced to one-eighth while retaining a noise suppression capability similar to that in the conventional method. A subjective evaluation experiment confirms that voice quality similar to that in the usual method is attained. (c) 2005 Wiley Periodicals, Inc.
Scientific journal, English - A robust NLMS algorithm for acoustic echo cancellation
S Shimauchi; Y Haneda; A Kataoka
ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE, SCRIPTA TECHNICA-JOHN WILEY & SONS, 89, 11, 1-9, 2006, Peer-reviwed, When an adaptive filter is applied for an acoustic echo canceler, realization of robustness in the presence of double talk is an important subject. In this paper, based on M-estimation, a robust statistical technique, a cost function expressing the normalized least-mean-square algorithm (NLMS), is modified to derive the gradient-limited NLMS (GL-NLMS) algorithm. This technique makes use of the statistical characteristic that the ratio of the error and reference signal levels has different distributions during single talk and double talk. The gradient used for updating in the NLMS algorithm is limited to a nonlinear form, so high robustness against double talk is achieved. (c) 2006 Wiley Periodicals, Inc.
Scientific journal, English - エリア内の再生特性を考慮した音の局所的再生
植松尚; 羽田陽一; 片岡 章俊
日本音響学会誌, 62, 2, 89-97, 2006, Peer-reviwed
Scientific journal, Japanese - An acoustic echo canceler with noise and echo reduction
S Sakauchi; Y Haneda; M Tanaka; J Sasaki; A Kataoka
ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE, SCRIPTA TECHNICA-JOHN WILEY & SONS, 88, 3, 21-31, 2005, Peer-reviwed, yAn acoustic echo canceler, needed for hands-free telecommunications in videoconferencing, has been realized with sufficient performance for use in a quiet meeting room. However, demand has been increasing for the use of hands-free teleconferencing in an environment with ambient noise such as conference calls in a noisy office or in a running automobile. In response to such demands, this paper proposes an echo canceler with acapability of simultaneously suppressing stationary ambient noise superposed on the speech and also residual echo, which cannot be eliminated by conventional echo cancelers because it is buried in noise. In this paper, the noise reduction and echo reduction process, based on the short-time spectrum amplitude estimation, is first explained. Then the results of objective evaluation of a test-fabricated echo canceler are presented. The evaluation results confirm that the echo can be reduced by about 30 dB and the ambient noise by about 15 dB even in an environment with ambient noise, allowing satisfactory hands-free conference calling to be realized. (C) 2004 Wiley Periodicals, Inc.
Scientific journal, English - Frequency domain adaptive algorithm with nonlinear function of error-to-reference ratio for double-talk robust echo cancellation
Suehiro Shimauchi; Yoichi Haneda; Akitoshi Kataoka
Acoustical Science and Technology, 26, 1, 8-15, Jan. 2005, Peer-reviwed, Several adaptive algorithms for robust echo cancellation use nonlinear reference and/or error functions. Most of them require time-variant threshold estimators, e.g., noise level estimators or double-talk detectors, since their nonlinearities have to be adjusted in response to changes in near-end noise or speech signal levels. We propose a new frequency domain adaptive algorithm: the gradient-limited fast least-mean-squares (GL-FLMS), in which the coefficients are updated by using a nonlinear function of the error scaled by the reference magnitude, i.e., the error-to-reference ratio (ERR). When the acoustic coupling level between loudspeaker and microphone is bounded, the ERR is also bounded in the case of single-talk, but may increase during double-talk. The GL-FLMS limits unexpected increases in the ERR with fixed thresholds and prevents divergence of the coefficients, while not neglecting updates to adjust when a large reference signal introduces a large error during single-talk.
Scientific journal, English - 音響エコーキャンセラのための学習同定法のロバスト化
島内末廣; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J88-A, 8, 926-934, 2005, Peer-reviwed
Scientific journal, Japanese - 過去の誤差信号を用いないアフィン射影アルゴリズムの実現とその音響エコーキャンセラへの適用
島内末廣; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, The Institute of Electronics, Information and Communication Engineers, J88-A, 11, 1235-1245, 2005, Peer-reviwed, 音響エコーキャンセラにおけるエコー経路推定アルゴリズムとして, しばしば採用されているアフィン射影アルゴリズム(APA : Affine Projection Algorithm)は, 現在及び過去に得られた所望信号とフィルタ出力との誤差信号に関する入出力関係を連立させて解くことにより, フィルタ更新計算を行い, 音声のような有色信号に対し, 学習同定法(NLMSアルゴリズム)と比べ, 収束速度が高いという特徴をもつ. 本論文では, 2次のAPAに対し, APAが用いる過去の誤差信号を, 現在と過去の入力信号の相関と現在の誤差信号とに基づく推定値で置き換えることにより, より少ない演算量で, APAの性質を近似的に実現し, 更に, 音声などの非定常信号入力時にAPAよりも高いロバスト性を有する新しい適応アルゴリズムとしてEMAPA(Error-Memoryless Affine Projection Algorithm)を提案する.
Scientific journal, Japanese - STSA推定に基づくエコー抑圧処理のゲイン強調化方式
阪内澄宇; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J88-A, 6, 695-703, 2005, Peer-reviwed
Scientific journal, Japanese - A solution to echo path imbalance problem in stereo echo cancellation
S Emura; Y Haneda; A Kataoka
2004 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOL IV, PROCEEDINGS, IEEE, 4, 129-132, 2004, Peer-reviwed, We propose a novel step-size control method that treats the echo path imbalance problem in stereo echo cancellation. The imbalance between two echo paths in magnitude slows down the convergence rate of the adaptive filter for stereo echo cancellation. To speed up the convergence rate, the proposed method extracts the information about filter-coefficient errors of each unknown echo path, and controls the step-size of each channel accordingly.
International conference proceedings, English - 雑音抑圧及びエコー抑圧機能を備えた音響エコーキャンセラ
阪内澄宇; 羽田陽一; 田中雅史; 佐々木潤子; 片岡章俊
電子情報通信学会論文誌 A, J87-A, 4, 448-457, 2004, Peer-reviwed
Scientific journal, Japanese - 処理遅延を削減した1チャネル雑音抑圧
阪内澄宇; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J87-A, 11, 1367-1375, 2004, Peer-reviwed
Scientific journal, Japanese - 方向別自動音量調整マイクロホンアレー (電気音響, 音響一般)
小林和則; 古家賢一; 羽田陽一; 片岡章俊
電子情報通信学会論文誌 A, J87-A, 12, 1491-1501, 2004, Peer-reviwed
Scientific journal, Japanese - Implementing and evaluating an audio teleconferencing terminal with noise and echo
S. Sakauchi; A. Nakagawa; Y. Haneda; A. Kataoka
The International Workshop on Acoustic Signal Enhancement, -, Sep. 2003, Peer-reviwed
International conference proceedings, English - A method of coherence-based step-size control for robust stereo echo cancellation
S Emura; Y Haneda
2003 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOL V, PROCEEDINGS, IEEE, 5, 592-595, 2003, Peer-reviwed, The fast adaptive algorithm is required for stereo echo cancellation because the strong inter-channel cross-correlation of stereo received signals leads to ill-conditioned normal equation to be solved by the adaptive filter. An adaptive algorithm for this task should also be robust against such disturbances as near-end speech and near-end noise. We propose a coherence-based method of step-size control that provides robustness in stereo echo cancellation. Computer simulation demonstrated that the method was robust against near-end speech and noise, and was capable of quickly tracking change in echo paths.
International conference proceedings, English - ステレオ信号間のパワー差を考慮したステレオエコーキャンセラに関する一検討
中川朗; 羽田陽一
電子情報通信学会論文誌 A, J86-A, 10, 989-997, 2003, Peer-reviwed
Scientific journal, Japanese - Active noise control with a virtual microphone based on common-acoustical-pole and residue model
Y Haneda
2002 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS I-IV, PROCEEDINGS, IEEE, 2, 1877-1880, 2002, Peer-reviwed, A local active noise control (ANC)system with a virtual microphone based on the common-acoustical-pole and residue (CAPR) model is proposed.
In conventional ANC systems, the microphone(s) must be set close to the listener's car for maximum effectiveness, because the "quiet zone" is generated only around the error sensor in a room. However, this is often cumbersome, so the microphoneshould ideally be located away from the target "quiet" position.
When modeling the room transfer functions (RTFs) based on the CAPR model, we can extrapolate an RTF from several RTFs observed in a simple rectangular room. By applying this extrapolation method to the ANC system, we can estimate and reduce the noise sound pressure at the virtual microphone position.
Computer simulation or the proposed ANC system in a small rectangular room demonstrates that the peak frequencies of the noise can be eliminated well at the virtual microphone position.
International conference proceedings, English - A study of an adaptive alogrithm for stereo signals with a power difference
A Nakagawa; Y Haneda
2002 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS I-IV, PROCEEDINGS, IEEE, 2, 1913-1916, 2002, Peer-reviwed, With broadband networks spreading, high-presence acoustic communication like stereophonic full-duplex communication is desired. When loudspeakers and microphones are used for hands-free communication, a stereophonic acoustic echo canceller (SAEC) is required to avoid howling and echoes. However, when there is a large power difference between stereo signals received from the far-end, the performance of the acoustic echo canceller is degraded; that is the convergence speed of the coefficient error is low. In this paper, to overcome this problem, we propose a new adaptive algorithm for the SAEC that uses the adjustment vector. Each channel received a signal normalized by the root of their squared sum. Computer simulation shows that the proposed algorithm improves the convergence speed for stereo signals with a power difference.
International conference proceedings, English - Enhanced frequency-domain adaptive algorithm for stereo echo cancellation
S Emura; Y Haneda; S Makino
2002 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS I-IV, PROCEEDINGS, IEEE, 2, 1901-1904, 2002, Peer-reviwed, Highly cross-correlated input signals create the problem of slow convergence of misalignment in stereo echo cancellation even after undergoing non-linear preprocessing. We propose a new frequency-domain adaptive algorithm that improves the convergence rate by increasing the contribution of non-linearity in the adjustment vector. Computer simulation showed that it is effective when the non-linearity gain is small.
International conference proceedings, English - Adaptive algorithm enhancing decorrelated additive signals for stereo echo cancellation
S. Emura; Y. Haneda; S. Makino
The International Workshop on Acoustic Signal Enhancement (IWAENC), 67-70, Sep. 2001, Peer-reviwed
International conference proceedings, English - 全指向性を持つスピーカ・マイクロホン一体型通話装置の設計
中川朗; 島内末廣; 羽田陽一; 青木茂明; 牧野 昭二
日本音響学会誌, 57, 8, 509-516, 2001, Peer-reviwed
Scientific journal, Japanese - Subjective assessment of the desired echo return loss for subband acoustic echo cancellers
S Sakauchi; Y Haneda; S Makino; M Tanaka; Y Kaneda
IEICE TRANSACTIONS ON FUNDAMENTALS OF ELECTRONICS COMMUNICATIONS AND COMPUTER SCIENCES, IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG, E83A, 12, 2633-2639, Dec. 2000, Peer-reviwed, We investigated the dependence of the desired echo return loss on frequency for various hands-free telecommunication conditions by subjective assessment The desired echo return loss as a function of frequency (DERLf) is all important factor in the design and performance evaluation of a subband echo canceller, and it is a measure of what is considered all acceptable echo caused by electrical loss in the transmission line. The DERLf during single talk was obtained as attenuated band-limited echo levels that subjects did not find objectionable when listening to the near-end speech and its band-limited echo under various hands-free telecommunication conditions. When we investigated the DERLf during double-talk, subjects also heard the speech in the far-end room from a loudspeaker. The echo was limited to a 250-Hz bandwidth assuming the use of a subband echo canceller. The test results showed that: (1) when the transmission delay was short (30 ms), the echo component around 2 to 3 kHz was the most objectionable to listeners. (2) as the transmission delay rose to 300 ms, the echo component around 1 kHz became the most objectionable; (3) when the room reverberation time was relatively long (about 500 ms). the echo cumyonent around 1 kHz was the most objectionable even if the transmission delay was short; and ( 1) the DERLf during double-talk was about 5 to 10dB lower than that during single-talk. Use of these DERLf values will enable the design of mure efficient subband echo cancellers.
Scientific journal, English - Channel-number-compressed multi-channel acoustic echo canceller for high-presence teleconferencing system with large display
A Nakagawa; S Shimauchi; Y Haneda; S Aoki; S Makino
2000 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, PROCEEDINGS, VOLS I-VI, IEEE, 2, 813-816, 2000, Peer-reviwed, Sound localization is important to make conversation easy between local and remote sites in a teleconference. This requires a multi-channel sound system having a multi-channel acoustic echo canceller (MAEC). The appropriate number of channels is determined from a trade-off between high presence and MAEC performance, so it is not possible to increase the channel number by much.
We propose a channel-number-compressed MAEC to provide teleconferencing systems that exhibit high presence. The channel number of the MAEC inputs is compressed and that of its outputs is expanded.
International conference proceedings, English - Common-acoustical-pole and residue model and its application to spatial interpolation and extrapolation of a room transfer function
Y Haneda; Y Kaneda; N Kitawaki
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 7, 6, 709-717, Nov. 1999, Peer-reviwed, A method is proposed for modeling a room transfer function (RTF) by using common acoustical poles and their residues. The common acoustical poles correspond to the resonance frequencies (eigenfrequencies) of the room, so they are independent of the source and receiver positions. The residues correspond to the eigenfunctions of the room. Therefore, the residue, which is a function of the source and receiver positions, can be expressed using simple analytical functions for rooms with a simple geometry such as rectangular. That is, the proposed model can describe RTF variations using simple residue functions. Based on the proposed common-acoustical-pole and residue model, methods are also proposed for spatially interpolating and extrapolating RTF's, Because the common acoustical poles are invariant in a given room, the interpolation or extrapolation of RTF's is reformulated as a problem of interpolating or extrapolating residue values. The experimental results for a rectangular room, in which the residue values are interpolated or extrapolated by using a cosine function or a linear prediction method, demonstrate that unknown RTF's can be well estimated at low frequencies from known (measured) RTF's by using the proposed methods.
Scientific journal, English - Subband echo canceler with an exponentially weighted stepsize NLMS adaptive filter
S Makino; Y Haneda
ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE, SCRIPTA TECHNICA-JOHN WILEY & SONS, 82, 3, 49-57, Mar. 1999, Peer-reviwed, This paper proposes a novel adaptive algorithm for an echo canceler. In this algorithm, the number of operations and memory capacity are equivalent to those of the conventional NLMS algorithm but the convergence speed is twice that using the conventional algorithm. This adaptive algorithm is referred to as subband ES (exponentially weighted stepsize). In the algorithm, the frequency bands of the received input signal and echo signal are divided into multiple subbands, and echo is independently canceled in each subband. Each adaptive filter in each subband has independent coefficients with an independent stepsize. The stepsize is time-independent and its weight is exponentially proportional to the change of the impulse response within the frequency region, such as the expected value of the difference between the waveforms of two impulse responses. As a result, the characteristic of the acoustic echo path in each frequency band is analyzed using the adaptive algorithm to improve the convergence characteristic. Using the results of computer simulation and experimental results obtained via an experimental setup with DSP, it is shown that the convergence speed with respect to input voice signal can be about 4 times faster when using echo cancellation based on the new algorithm than in conventional full-band echo cancellation based on the NLMS algorithm. (C) 1998 Scripta Technica, Electron Comm Jpn Pt 3, 82(3): 49-57, 1999.
Scientific journal, English - Common-acoustical-pole and zero modeling of head-related transfer functions
Y Haneda; S Makino; Y Kaneda; N Kitawaki
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 7, 2, 188-196, Mar. 1999, Peer-reviwed, Use of a common-acoustical-pole and zero model is proposed for modeling head-related transfer functions (HRTF's) for various directions of sound incidence. The HRTF's are expressed using the common acoustical poles, which do not depend on the source directions, and the zeros, which do, The common acoustical poles are estimated as they are common to HRTF's for various source directions; the estimated values of the poles agree well with the resonance frequencies of the ear canal. Because this model uses only the zeros to express the HRTF variations due to changes in source direction, it requires fewer parameters (the order of the zeros) that depend on the source direction than do the conventional all zero or pole/zero models. Furthermore, the proposed model can extract the zeros that are missed in the conventional models because of pole-zero cancellation. As a result, the directional dependence of the zeros can be traced well. Analysis of the zeros for HRTF's on the horizontal plane showed that the nonminimum-phase zero variation was well formulated using a simple pinna-reflection model, The common-acoustical-pole and zero (CAPZ) model is thus effective for modeling and analyzing HRTF's.
Scientific journal, English - A stereo echo canceller implemented using a stereo shaker and a duo-filter control system
S Shimauchi; S Makino; Y Haneda; A Nakagawa; S Sakauchi
ICASSP '99: 1999 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, PROCEEDINGS VOLS I-VI, IEEE, 2, 857-860, 1999, Peer-reviwed, Stereo echo cancellation has been achieved and used in daily teleconferencing. To overcome the non-uniqueness problem, a stereo shaker is introduced in eight frequency bands and adjusted so as to be inaudible and not affect stereo perception. A duo-filter control system including a continually running adaptive filter and a fixed filter is used for double-talk control. A second-order stereo projection algorithm is used in the adaptive filter. A stereo voice switch is also included. This stereo echo canceller was tested in two-way conversation in a conference room, and the strength of the stereo shaker was subjectively adjusted. A misalignment of 20 dB was obtained in the teleconferencing environment, and changing the talker's position in the transmission room did not affect the cancellation. This echo canceller is now used daily in a high-presence teleconferencing system and has been demonstrated to more than 300 attendees.
International conference proceedings, English - New configuration for a stereo echo canceller with nonlinear pre-processing
S Shimauchi; Y Haneda; S Makino; Y Kaneda
PROCEEDINGS OF THE 1998 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, VOLS 1-6, IEEE, 6, 3685-3688, 1998, Peer-reviwed, A new configuration for a stereo echo canceller with nonlinear pre-processing is proposed. The pre-processor which adds uncorrelated components to the original received stereo signals improves the adaptive filter convergence even in the conventional configuration. However, because of the inaudibility restriction, the preprocessed signals still include a large amount of the original stereo signals which are often highly cross-correlated. Therefore, the improvement is limited. To overcome this, our new stereo echo canceller includes exclusive adaptive filters whose inputs are the uncorrelated signals generated in the pre-processor. These exclusive adaptive filters converge to true solutions without suffering from cross-correlation between the original stereo signals. This is demonstrated through computer simulation results.
International conference proceedings, English - Interpolation and extrapolation of room transfer functions based on common acoustical poles and their residues
Y. Haneda; Y. Kaneda
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), Mon-A-1, Oct. 1997, Peer-reviwed
International conference proceedings, English - Subband acoustic echo canceller using two different analysis filters and 8th order projection
A Nakagawa; Y Haneda; S Makino
The International Workshop on Acoustic Signal Enhancement (IWAENC), -, -, Sep. 1997, Peer-reviwed
International conference proceedings, English - Multiple-point equalization of room transfer functions by using common acoustical poles
Y Haneda; S Makino; Y Kaneda
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 5, 4, 325-333, Jul. 1997, Peer-reviwed, A multiple-point equalization filter using the common acoustical poles of room transfer functions is proposed, The common acoustical poles correspond to the resonance frequencies, which are independent of source and receiver positions. They are estimated as common autoregressive (AR) coefficients from multiple room transfer functions. The equalization is achieved with a finite impulse response (FIR) filter, which has the inverse characteristics of the common acoustical pole function. Although the proposed filter cannot recover the frequency response dips of the multiple room transfer functions, it can suppress their common peaks due to resonance; it is also less sensitive to changes in receiver position, Evaluation of the proposed equalization filter using measured room transfer functions shows that it can reduce the deviations in the frequency characteristics of multiple room transfer functions better than a conventional multiple-point inverse filter, Experiments show that the proposed filter enables 1-5 dB additional amplifier gain in a public address system without acoustic feedback at multiple receiver positions, Furthermore, the proposed filter reduces the reflected sound in room impulse responses without the pre-echo that occurs with a multiple-point inverse filter. A multiple-point equalization filter using common acoustical poles can thus equalize multiple room transfer functions by suppressing their common peaks.
Scientific journal, English - Subband stereo echo canceller using the projection algorithm with fast convergence to the true echo path
S Makino; K Strauss; S Shimauchi; Y Haneda; A Nakagawa
1997 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS I - V, IEEE COMPUTER SOC, 1, 299-302, 1997, Peer-reviwed, This paper proposes a new subband stereo echo canceller that converges to the true echo path impulse response much faster than conventional stereo echo cancellers. Since signals are bandlimited and downsampled in the subband structure, the time interval between the subband signals become longer, so the variation of the crosscorrelation between the stereo input signals becomes large. Consequently, convergence to the true solution is improved. Furthermore, the projection algorithm, or affine projection algorithm, is applied to further speed up the convergence. Computer simulations using stereo signals recorded in a conference room demonstrate that this method significantly improves convergence speed and almost solves the problem of stereo echo cancellation with low computational load.
International conference proceedings, English - Implementation and evaluation of an acoustic echo canceller using duo-filter control system
Y. Haneda; S. Makino; J. Kojima; S. Shimauchi
European Signal Processing Conference (EUSIPCO), 1115, 1, 1115-1118, Sep. 1996, Peer-reviwed
International conference proceedings, English - SSB subband echo canceller using low-order projection algorithm
S Makino; J Noebauer; Y Haneda; A Nakagawa
1996 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, CONFERENCE PROCEEDINGS, VOLS 1-6, IEEE, 2, 945-948, 1996, Peer-reviwed
International conference proceedings, English - 周波数帯域における音響エコー経路の変動特性を反映させたサブバンドESアルゴリズム
牧野昭二; 羽田陽一
電子情報通信学会論文誌 A, The Institute of Electronics, Information and Communication Engineers, J79-A, 6, 1138-1146, 1996, Peer-reviwed, 本論文は, 従来のNLMSアルゴリズムと同等の演算量と記憶容量で収束速度が約2倍の, 新しいエコーキャンセラ用適応アルゴリズムを提案するものである. サブバンド ES (exponentially weighted stepsize) アルゴリズムと名づけたこの適応アルゴリズムでは, 受話入力信号とエコー信号を複数の周波数帯域に分割し, それぞれの周波数帯域で独立にエコーを消去するサブバンドエコーキャンセラにおいて, それぞれの周波数帯域に設けた適応形トランスバーサルフィルタのそれぞれの係数に対して, 異なるステップサイズを用いている. これらのステップサイズは時不変で, その周波数帯域における室内インパルス応答の変化分, 例えば二つのインパルス応答波形の差, の期待値に比例して指数的に重み付けられている. その結果, 各周波数帯域における音響エコー経路の変動特性の違いを適応アルゴリズムに反映させ, 収束特性を改善することができる. ここでは, 室内音場のインパルス応答データを用いた計算機シミュレーション, およびDSPで構成した実験装置を用いた実時間評価実験を行い, NLMSアルゴリズムを用いた従来のフルバンドエコーキャンセラに比べて, 音声入力に対する収束速度を約4倍にできることを明らかにする.
Scientific journal, Japanese - Common Acoustical Pole and Zero Modeling of Room Transfer Functions
Yoichi Haneda; Shoji Makino; Yutaka Kaneda
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC, 2, 2, 320-328, Apr. 1994, Peer-reviwed, A new model for a room transfer function (RTF) by using common acoustical poles that correspond to resonance properties of a room is proposed. These poles are estimated as the common values of many RTF's corresponding to different source and receiver positions. Since there is one-to-one correspondence between poles and AR coefficients, these poles are calculated as common AR coefficients by two methods: i) using the least squares method, assuming all the given multiple RTF's have the same AR coefficients and ii) averaging each set of AR coefficients estimated from each RTF. The estimated poles agree well with the theoretical poles when estimated with the same order as the theoretical pole order. When estimated with a lower order than the theoretical pole order, the estimated poles correspond to the major resonance frequencies, which have high Q factors. Using the estimated common AR coefficients, the proposed method models the RTF's with different MA coefficients. This model is called the common-acoustical-pole and zero (CAPZ) model, and it requires far fewer variable parameters to represent RTF's than the conventional all-zero or pole/zero model. This model was used for an acoustic echo canceller at low frequencies, as one example. The acoustic echo canceller based on the proposed model requires half the variable parameters and converges 1.5 times faster than one based on the all-zero model, confirming the efficiency of the proposed model.
Scientific journal, English - Common acoustical poles independent of sound directions and modeling of head-related transfer functions
Yoichi Haneda; Shoji Makino; Yutaka Kaneda
Journal of the Acoustical Society of Japan (E) (English translation of Nippon Onkyo Gakkaishi), 15, 4, 277-279, 1994, Peer-reviwed
Scientific journal, English - Arma modeling of a room transfer function at low frequencies
Yoichi Haneda; Shoji Makino; Yutaka Kaneda; Nobuo Koizumi
Journal of the Acoustical Society of Japan (E) (English translation of Nippon Onkyo Gakkaishi), 15, 5, 353-355, 1994, Peer-reviwed
Scientific journal, English - MODELING OF A ROOM TRANSFER-FUNCTION USING COMMON ACOUSTICAL POLES
Y HANEDA; S MAKINO; Y KANEDA
ICASSP-92 - 1992 INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS 1-5, I E E E, 2, B213-B216, 1992, Peer-reviwed
International conference proceedings, English
MISC
- インタラクティブオーディオに向けた 焦点音源のリアルタイム生成
羽田陽一
Lead, Oct. 2023, 日本音響学会誌, 79, 10, 502-509, Japanese, Invited, Introduction scientific journal - 会長就任にあたって
羽田陽一
Lead, Jun. 2023, 日本音響学会誌, 79, 6, 291-292, Japanese, Invited, Introduction scientific journal - 空間フーリエ変換を用いたアレイ信号処理
羽田 陽一
日本音響学会, 01 Feb. 2018, 日本音響学会誌, 74, 2, 74-82, Japanese, Peer-reviwed, Introduction scientific journal - 音の波数領域信号処理
羽田陽一
電子情報通信学会, 2018, 電子情報通信学会 基礎・境界ソサイエティ Fundamentals Review, 11, 4, 243-255, Japanese, Peer-reviwed, Introduction scientific journal - Design of a directive loudspeaker array filter based on weighted least-squares method by controlling window function for desired sound pressure
佐藤 航也; 関 貴志; 羽田 陽一
電子情報通信学会, 28 Mar. 2016, 電子情報通信学会技術研究報告 = IEICE technical report : 信学技報, 115, 522, 219-224, Japanese, 0913-5685, 40020792043, AA11943613 - Directivity control for a regular icosahedron loudspeaker based on the adaptive weighted least-squares method
大小原 亮; 羽田 陽一
電子情報通信学会, 28 Mar. 2016, 電子情報通信学会技術研究報告 = IEICE technical report : 信学技報, 115, 522, 213-218, Japanese, 0913-5685, 40020792034, AA11943613 - Virtual Extrapolation of Microphone Array Length for Two Sound Sources Visualization
マウリン; 羽田 陽一
電子情報通信学会, 28 Mar. 2016, 電子情報通信学会技術研究報告 = IEICE technical report : 信学技報, 115, 522, 201-206, Japanese, 0913-5685, 40020792009, AA11943613 - A study on transaural system using directivity control of regular icosahedron loudspeaker array
大小原 亮; 羽田 陽一
電子情報通信学会, 03 Jul. 2015, 電子情報通信学会技術研究報告 = IEICE technical report : 信学技報, 115, 126, 1-6, Japanese, 0913-5685, 40020557459, AN10164817 - 音源位置事前情報を用いた超解像型音場収音・再現における空間相関分布の導入
小山翔一; 古家賢一; 羽田陽一; 猿渡洋
06 Mar. 2015, 日本音響学会研究発表会講演論文集(CD-ROM), 2015, ROMBUNNO.3-10-13, Japanese, 1880-7658, 201502223763473099 - Investigation of local area sound reproduction using a circular array of loudspeakers with null-space based sound field control
SEKI Takayuki; HANEDA Yoichi
We investigated a local area sound reproduction system where the listener can hear the sound only in the proximity of the loudspeaker. In this paper, we used the circular loudspeaker array with null-space based sound field control (NBSFC) to create a local area sound reproduction system based on the boundary surface control principle. The performance of the prototype system of the circular loudspeaker array was evaluated in an anechoic chamber. As a result, the prototype system exhibited a quicker sound attenuation than the monopole sound source in the frequency range of 500-1500 Hz. In addition, we compared the filter coefficients of the loudspeaker array of the NBSFC method with those of the evanescent wave method. In the NBSFC method, the filters on both side of the filter that has a positive phase have opposite phases and half amplitudes. On the other hand, in the evanescent wave method, the filters alternately exhibit a plus or minus sign while having the same amplitude. In both cases, the sounds radiated from the loudspeakers may instantaneously cancel each other by opposite phases to achieve a quicker sound attenuation., The Institute of Electronics, Information and Communication Engineers, 02 Mar. 2015, IEICE technical report. Signal processing, 114, 474, 231-236, Japanese, 0913-5685, 110010017490, AA11943613 - Auditory Manipulation of Sound Field-Reality and Individuality-
武田一哉; 西野隆典; 丹羽健太; 羽田陽一; 猿渡洋; 西村竜一
2015, 電子情報通信学会技術研究報告, 115, 126(EA2015 1-9), 0913-5685, 201502214131742142 - 球面調和関数展開による音圧内挿を用いた超接話球面マイクロホンアレイ
羽田陽一; 古家賢一; 小山翔一; 丹羽健太
17 Sep. 2013, 日本音響学会研究発表会講演論文集(CD-ROM), 2013, ROMBUNNO.1-1-15, Japanese, 1880-7658, 201302201860563827 - 音源位置事前情報を用いた音圧―駆動信号変換に基づく音場再現
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一
17 Sep. 2013, 日本音響学会研究発表会講演論文集(CD-ROM), 2013, ROMBUNNO.3-1-2, Japanese, 1880-7658, 201302251172616892 - 伝達特性間の独立性を用いた拡散センシング型アレーの構造評価
丹羽健太; 日岡裕輔; 小林和則; 古家賢一; 羽田陽一
05 Mar. 2013, 日本音響学会研究発表会講演論文集(CD-ROM), 2013, ROMBUNNO.2-10-7, Japanese, 1880-7658, 201302229305619517 - 球・円状マイクロホン・スピーカアレイのための波面再構成フィルタ
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一
05 Mar. 2013, 日本音響学会研究発表会講演論文集(CD-ROM), 2013, ROMBUNNO.1-10-5, Japanese, 1880-7658, 201302243712382532 - 円調和ビームフォーミングによる波面再構成フィルタの残響環境下性能改善
小山翔一; LEE Timothy; 古家賢一; 日和崎祐介; 羽田陽一
05 Mar. 2013, 日本音響学会研究発表会講演論文集(CD-ROM), 2013, ROMBUNNO.1-P-19, Japanese, 1880-7658, 201302288743501061 - Frequency domain implementation of cascaded nonlinear adaptive filter with hard clipping function
島内 末廣; 大室 仲; 羽田 陽一
[電子情報通信学会信号処理研究専門委員会], 28 Nov. 2012, 信号処理シンポジウム講演論文集, 27, 531-536, Japanese, 1881-4654, 40019803031 - 高解像度映像に対するアンビエント5.1ch再生方式の主観評価実験分析結果
清原健司; 古家賢一; 木全英明; 田中康暁; 深澤勝彦; 羽田陽一; 小島明
11 Sep. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.1-9-8, Japanese, 1880-7658, 201202298206879293 - 最大SN比法への拡散センシングの適用
丹羽健太; 日岡裕輔; 荒木章子; 古家賢一; 羽田陽一
11 Sep. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.1-P-15, Japanese, 1880-7658, 201202207673659595 - 円筒状マイクロホン・スピーカアレーのための波面再構成フィルタ
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一; 鈴木陽一
11 Sep. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.1-9-13, Japanese, 1880-7658, 201202221411605314 - 拡散センシングに基づく指向制御~雑音出力パワーに対するフィルタ長の影響~
丹羽健太; 日岡裕輔; 古家賢一; 羽田陽一
11 Sep. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.2-9-7, Japanese, 1880-7658, 201202233699825036 - 波数領域における残留エコー消去方法
江村暁; 小山翔一; 古家賢一; 羽田陽一
11 Sep. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.3-4-4, Japanese, 1880-7658, 201202249213549583 - エバネッセント波の減衰を模擬した音場制御型エリア再生
伊藤弘章; 古家賢一; 羽田陽一
06 Mar. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.1-Q-2, Japanese, 1880-7658, 201202217641409622 - スピーカ指向性を考慮した時空間周波数領域での音場再現フィルタ設計手法
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一
06 Mar. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.2-Q-6, Japanese, 1880-7658, 201202277926748296 - 残響音の等方到来を仮定した直間比推定
日岡裕輔; 古家賢一; 丹羽健太; 羽田陽一
06 Mar. 2012, 日本音響学会研究発表会講演論文集(CD-ROM), 2012, ROMBUNNO.1-Q-4, Japanese, 1880-7658, 201202278803659880 - 反射率可変の壁面音響システムを用いた音場再生システム
岩永景一郎; 岩永景一郎; 尾本章; 河原一彦; 古家賢一; 清原健司; 岡本学; 羽田陽一
13 Sep. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.3-5-14, Japanese, 1880-7658, 201102217025087029 - 大型多重反射板付きマイクロホンアレーを用いた超指向性収音
丹羽健太; 阪内澄宇; 古家賢一; 岡本学; 羽田陽一
13 Sep. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.1-4-5, Japanese, 1880-7658, 201102244162749214 - 音場収音・再現のための時空間周波数領域信号変換法
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一
13 Sep. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.2-4-1, Japanese, 1880-7658, 201102253672426931 - 円形スピーカアレーを用いたエバネッセント波再生手法について
伊藤弘章; 古家賢一; 羽田陽一
13 Sep. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.1-P-2, Japanese, 1880-7658, 201102260616388594 - 時空間スペクトルの位相シフトによる音場再現位置の制御
小山翔一; 古家賢一; 日和崎祐介; 羽田陽一
13 Sep. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.2-4-2, Japanese, 1880-7658, 201102291268566761 - Picking up sounds at different distances with microphone array using reflected sounds
丹羽健太; 阪内澄宇; 古家賢一; 岡本学; 羽田陽一
16 Jun. 2011, 電子情報通信学会技術研究報告, 111, 89(EA2011 30-42), 31-36, Japanese, 0913-5685, 201102218714879630 - Study on super directive microphone array using multiple reflected sounds
丹羽健太; 阪内澄宇; 古家賢一; 岡本学; 羽田陽一
16 Jun. 2011, 電子情報通信学会技術研究報告, 111, 89(EA2011 30-42), 25-30, Japanese, 0913-5685, 201102273244696941 - A study of acoustical evanescent wave reproduction using planar array
伊藤弘章; 古家賢一; 羽田陽一
11 Mar. 2011, 電子情報通信学会技術研究報告, 110, 471(EA2010 126-134), 29-34, Japanese, 0913-5685, 201102257150877599 - 球調和関数展開に基づく多重極音源を用いた任意の指向性制御
古家賢一; 羽田陽一; 尾本章; 河原一彦
02 Mar. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.3-9-17, Japanese, 1880-7658, 201102204038635543 - 角度スペクトル微分を用いた波面合成法の実音場評価
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
02 Mar. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.3-9-6, Japanese, 1880-7658, 201102236583899280 - 狭指向性を実現する反射板利用型マイクロホンアレー
丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
02 Mar. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.1-9-18, Japanese, 1880-7658, 201102255980733643 - 直線状スピーカアレーを用いたエバネッセント波再生手法について
伊藤弘章; 古家賢一; 羽田陽一
02 Mar. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.3-P-57(D), Japanese, 1880-7658, 201102259890363648 - 球調和関数展開に基づく多重極音源型エンドファイアスピーカアレー
羽田陽一; 古家賢一; 伊藤弘章; 尾本章; 河原一彦
02 Mar. 2011, 日本音響学会研究発表会講演論文集(CD-ROM), 2011, ROMBUNNO.3-9-18, Japanese, 1880-7658, 201102296267557370 - エリア再生のための多点制御法による音波の減衰制御
伊藤弘章; 古家賢一; 羽田陽一
28 Feb. 2011, 電子情報通信学会大会講演論文集, 2011, 174, Japanese, 1349-1369, 201102268585506500 - 多重反射を利用した超指向性収音技術の検討
丹羽健太; 阪内澄宇; 古家賢一; 岡本学; 羽田陽一
2011, 音楽音響研究会資料, 30, 2, 25-30, Japanese, 0912-7283, 201102232683401089 - 反射音を利用したマイクロホンアレーによる距離別収音技術
丹羽健太; 阪内澄宇; 古家賢一; 岡本学; 羽田陽一
2011, 音楽音響研究会資料, 30, 2, 31-36, Japanese, 0912-7283, 201102282585702655 - 逆伝播演算による波面合成位置の操作
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
07 Sep. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.3-Q-6, Japanese, 1880-7658, 201002218293169975 - エバネッセント波を用いたエリア再生に関する研究
伊藤弘章; 古家賢一; 羽田陽一
07 Sep. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.2-P-18, Japanese, 1880-7658, 201002234712418862 - 最小分散ビームフォーマを用いた空間周波数解析による反射音情報推定
丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
07 Sep. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.2-P-19, Japanese, 1880-7658, 201002255237415338 - 伝達経路の時間応答特性を考慮した相関関数推定法と残響除去への適用
江村暁; 古家賢一; 羽田陽一
07 Sep. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.2-10-5, Japanese, 1880-7658, 201002269874955207 - 角度スペクトル微分による音圧勾配取得に基づく波面合成法
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
07 Sep. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.1-10-9, Japanese, 1880-7658, 201002272270378709 - 線状スピーカアレーを用いたエリア再生について
伊藤弘章; 古家賢一; 羽田陽一
01 Mar. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.2-P-24, Japanese, 1880-7658, 201002221739188627 - 二次元格子状マイクロホンアレーを用いた反射音情報の推定
丹羽健太; 日岡裕輔; 阪内澄宇; 古家賢一; 羽田陽一
01 Mar. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.2-5-7, Japanese, 1880-7658, 201002227479660970 - 高臨場遠隔通信のためのサウンドウォールシステム
阪内澄宇; 古家賢一; 小山翔一; 羽田陽一
01 Mar. 2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, ROMBUNNO.3-P-30, Japanese, 1880-7658, 201002262617829395 - Multi-layer Speech Coding Using Combination of Modes Based on Sparseness of Input Spectrum : ITU-T G.722/G.711.1 Super Wideband Extension Candidate
FUKUI Masahiro; TSUTSUMI Kimitaka; SASAKI Shigeaki; HIWASAKI Yusuke; HANEDA Yoichi
This paper describes a coding algorithm for 8-14 kHz bandwith layer of a scalable speech codec, submitted by a consortium including NTT as a candidate in the qualification phase of ITU-T G.722/G.711.1 super wideband extension standardization. The candidate codec achieves high speech quality irrespective of the sound source by using the proposed multistage quantization in the MDCT domain that selects the best encoding scheme among four modes according to the characteristic of various sound sources. The subjective test results showed that the candidate codec obtained higher scores than the reference codec in all samples and passed all requirements of the standardization in listening quality., The Institute of Electronics, Information and Communication Engineers, 14 Jan. 2010, IEICE technical report, 109, 376, 279-284, Japanese, 0913-5685, 110008002498, AN10476092 - 多段エコー推定による多チャネルエコー消去法
江村暁; 羽田陽一
2010, 日本音響学会研究発表会講演論文集(CD-ROM), 2010, 1880-7658, 201002257498848390 - On relationship between speech bandwidth and word intelligibility in noisy environment
KURIHARA Sachiko; HIWASAKI Yusuke; SASAKI Shigeaki; HANEDA Yoichi
To quantify the relationship between intelligibility and frequency bandwidth of speech in transmission systems, word intelligibility tests were performed. To simulate the real communication models, ambient noises were added either on sending or receiving side and were compared in terms of frequency bandwidths(band-limited to 3.4, 7, and 15 kHz). We confirmed that wideband(7 and 15 kHz)speech was more intelligible than narrowband speech(3.4 kHz), and found that this results exhibit stronger tendency in less-familiar word sets., The Institute of Electronics, Information and Communication Engineers, 14 Dec. 2009, IEICE technical report, 109, 355, 225-230, 0913-5685, 110008002122, AN10091225 - On relationship between speech bandwidth and word intelligibility in noisy environment
KURIHARA Sachiko; HIWASAKI Yusuke; SASAKI Shigeaki; HANEDA Yoichi
To quantify the relationship between intelligibility and frequency bandwidth of speech in transmission systems, word intelligibility tests were performed. To simulate the real communication models, ambient noises were added either on sending or receiving side and were compared in terms of frequency bandwidths (band-limited to 3.4, 7, and 15 kHz). We confirmed that wideband (7 and 15 kHz) speech was more intelligible than narrowband speech (3.4 kHz), and found that this results exhibit stronger tendency in less-familiar word sets., The Institute of Electronics, Information and Communication Engineers, 14 Dec. 2009, IEICE technical report, 109, 356, 225-230, 0913-5685, 110008000962, AN10013221 - 音源向きと空間相関行列の固有空間の関連性
丹羽健太; 日岡裕輔; 阪内澄宇; 古家賢一; 羽田陽一
08 Sep. 2009, 日本音響学会研究発表会講演論文集(CD-ROM), 2009, ROMBUNNO.3-Q-22, 1880-7658, 200902237136615205 - 受音信号の直間比に基づく距離別収音の検討
日岡裕輔; 丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
08 Sep. 2009, 日本音響学会研究発表会講演論文集(CD-ROM), 2009, ROMBUNNO.2-4-11, 1880-7658, 200902239436817309 - 空間相関行列の固有値の比率に着目した発話者向きの推定
丹羽健太; 阪内澄宇; 岡本学; 古家賢一; 羽田陽一
10 Mar. 2009, 日本音響学会研究発表会講演論文集(CD-ROM), 2009, ROMBUNNO.3-P-8, 1880-7658, 200902206080500083 - A-4-18 ACCURATE ECHO POWER ESTIMATION FOR ECHO REDUCTION
Fukui Masahiro; Nakagawa Akira; Shimauchi Suehiro; Haneda Yoichi; Kataoka Akitoshi
The Institute of Electronics, Information and Communication Engineers, 04 Mar. 2009, Proceedings of the IEICE General Conference, 2009, 122-122, 110007127491, AN10471452 - Field Evaluation of the Information Assurance System by Automatic Speech Recognition at Office Meeting
織田修平; 水島昌英; 古家賢一; 政瀧浩和; 羽田陽一
13 Feb. 2009, 電子情報通信学会技術研究報告, 108, 435(WIT2008 56-68), 11-16, 0913-5685, 200902208585009166 - Evaluation of Function to Support Cooperation between Speakers and a Mender in the Information Assurance System by Automatic Speech Recognition
水島昌英; 織田修平; 政瀧浩和; 古家賢一; 羽田陽一
13 Feb. 2009, 電子情報通信学会技術研究報告, 108, 435(WIT2008 56-68), 17-22, 0913-5685, 200902217375091500 - 残響と雑音が存在する環境下での強調処理音声の品質評価
古家賢一; 羽田陽一; 片岡章俊
03 Sep. 2008, 日本音響学会研究発表会講演論文集(CD-ROM), 2008, ROMBUNNO.3-8-11, 1880-7658, 200902250771067989 - Emphasis of sounds in a specific 2-dimentional area using estimated positions of sound sources
日岡裕輔; 小林和則; 古家賢一; 羽田陽一; 片岡章俊
20 Jun. 2008, 電子情報通信学会技術研究報告, 108, 115(EA2008 26-36), 7-12, 0913-5685, 200902254417310090 - Stereo echo cancellation using MS stereo signals for input signals
NAKAGAWA Akira; HANEDA Yoichi; KATAOKA Akitoshi
The convergence speed of stereo echo cancellation is slower than that of monaural echo cancellation becouse of the strong correlation between stereo signals. However, stereo echo cancellation requires a fast convergence. The proposed method can improve the convergence speed by using additional and subtracted signals from the stereo signals as input signals for stereo echo cancellation. First, we show that convergence speed is the same, even if the additional and subtracted signals are used instead of the stereo signals. Next, we focus on the power difference between the additional and subtracted signals and improve the convergence speed by using a different normalization for each input signal. The power difference between stereo signals depends on the method of picking up speech, a hybrid configuration that properly uses the appropriate adaptive altorithm for various power differences is presented. The effectiveness of this is confirmed by computer simulations., The Institute of Electronics, Information and Communication Engineers, 30 May 2008, IEICE technical report, 108, 71, 27-32, 0913-5685, 110006825151, AA11943613 - Noise power estimation based on noise ratio in signal and its noise reduction performance
FUKUI Masahiro; SHIMAUCHI Suehiro; NAKAGAWA Akira; HANEDA Yoichi; KATAOKA Akitoshi
Noise power estimation for noisy speech signals is required for noise reduction systems. This paper presents a robust noise power estimation method. The proposed method can estimate the noise power even in a speech period with high accuracy by estimating the ratio of input signal power to noise power for every frequency bin successively. First, proposed method estimates the input-signal-power-to-noise-power ratio to calculate the noise power by assuming that the noise amplitude of every frequency bin is constant in a short period. Next, proposed method improves the estimation accuracy by compensating for the estimation error caused by time variations in the actual noise amplitude. The performance of these two methods is demonstrated by simulation results in which the noise was suppressed with high accuracy., The Institute of Electronics, Information and Communication Engineers, 07 Mar. 2008, IEICE technical report, 107, 532, 85-90, 0913-5685, 110006952148, AN10164817 - 音源位置の推定情報を用いた特定の2次元領域内の強調収音
日岡裕輔; 小林和則; 古家賢一; 羽田陽一; 片岡章俊
2008, 音楽音響研究会資料, 27, 2, 7-12, 0912-7283, 200902239696779842 - 多チャネルオーディオ通信におけるエコーキャンセラ技術
島内末廣; 江村 暁; 羽田陽一; 片岡 章俊
2008, 電子情報通信学会誌, 89, 12, 1073-1078, 0913-5693, 200902202089423035 - Accurate sound source localization using multiple small microphone arrays and ultrasonic distance measurement
小林和則; 古家賢一; 羽田陽一; 片岡章俊
30 Nov. 2007, 電子情報通信学会技術研究報告, 107, 370(EA2007 86-94), 13-18, 0913-5685, 200902251270007389 - Howling Canceller Based on Sparseness of Speech for Hands-Free System.
小林和則; 古家賢一; 羽田陽一; 片岡章俊
20 Jul. 2007, 電子情報通信学会技術研究報告, 107, 170(EA2007 35-38), 1-6, 0913-5685, 200902206110158423 - 次世代ネットワークを利用したプラットフォーム・アプリケーション技術 (特集 NGNが提供する新しいコミュニケーションとそれを支える技術)
川添雄彦; 柿沼隆馬; 羽田陽一
電気通信協会, 2007, NTT技術ジャ-ナル, 19, 4, 44-49, 0915-2318, 40015441465, AN1013019X - Toward High-quality and High-reality Teleconferencing
Y. Haneda
2007, NTT Technical Review, 4, 2, 6-11 - スピーカの非線形性に対応可能なマイクロホン対を用いた音響エコーキャンセラ
小林和則; 古家賢一; 羽田陽一; 片岡章俊
06 Sep. 2006, 日本音響学会研究発表会講演論文集(CD-ROM), 2006, 1-1-15, 1880-7658, 200902208058708200 - Reducing Estimation Error in Microphone and Sound Source Localization-An Approach to Solve Local Minimum Problem-
小林和則; 古家賢一; 羽田陽一; 片岡章俊
16 Jun. 2006, 電子情報通信学会技術研究報告, 106, 125(EA2006 20-24), 19-24, 0913-5685, 200902230814834210 - Single-channel non-stationary noise reduction for teleconferences
野口賢一; 阪内澄宇; 古家賢一; 羽田陽一; 片岡章俊
10 Nov. 2005, 電子情報通信学会技術研究報告, 105, 403(EA2005 68-73), 31-36, 0913-5685, 200902277333860930 - マイクロホンと音源位置の同時推定―1つ以上のマイクロホン間距離が既知の場合―
小林和則; 古家賢一; 羽田陽一; 片岡章俊
20 Sep. 2005, 日本音響学会研究発表会講演論文集(CD-ROM), 2005, 2-2-15, 1880-7658, 200902213868176798 - A Multi-Channel Echo Reduction by Send Signal Selection in Frequency Domain
FUKUI Masahiro; SHIMAUCHI Suehiro; HANEDA Yoichi; KATAOKA Akitoshi
08 Mar. 2005, 日本音響学会研究発表会講演論文集, 2005, 1, 511-512, 1340-3168, 10018038303, AN00351181 - Convergence analysis of the enhanced NLMS for stereo echo cancellation
EMURA Satoru; HANEDA Youichi; KATAOKA Akitoshi
08 Mar. 2005, 日本音響学会研究発表会講演論文集, 2005, 1, 509-510, 1340-3168, 10018038297, AN00351181 - AS-4-3 On Echo Cancellation for High Presence Telecommunication : Beyond Stereo to Multi-Channel
Shimauchi Suehiro; Emura Satoru; Haneda Yoichi; Kataoka Akitoshi
The Institute of Electronics, Information and Communication Engineers, 07 Mar. 2005, Proceedings of the IEICE General Conference, 2005, "S-65"-"S-66", 110004738146, AN10471452 - Echo Reduction for FM-bandwidth signal : Acoustic coupling estimation during double-talk periods
SAKAUCHI Sumitaka; HANEDA Yoichi; KATAOKA Akitoshi
Echo reduction based on short-time spectral amplitude estimation is nonlinear processing in the frequency domain for hands-free telecommunications. A frequency characteristic of acoustic coupling between loud-speaker and microphone is needed to calculate the echo-reducing gain. It is difficult to estimate the acoustic coupling during the double-talk period by the conventional method. Moreover, the acoustic coupling estimation speed at high frequencies is too slow. We propose new acoustic coupling estimation methods used in the echo reduction for FM-bandwidth (32 kHz sampling) signal. One using harmonic structures of echoes and near-end speech can estimate the acoustic coupling during double-talk periods. The other using frequency characteristics of an acoustic echo path can estimate the acoustic coupling around high frequencies faster than the conventional method. Computer simulation results demonstrate that the proposed method increased the estimation speed of the acoustic coupling around high frequencies during the double-talk period., The Institute of Electronics, Information and Communication Engineers, 21 Jan. 2005, Technical report of IEICE. EA, 104, 615, 9-14, 0913-5685, 110003284927, AN10164817 - A study on an error-memoryless affine projection algorithm : Application to acoustic echo cancellation
SHIMAUCHI Suehiro; HANEDA Yoichi; KATAOKA Akitoshi
The affine projection algorithm (APA) is one of the most popular adaptive algorithms for acoustic echo cancellation. The advantage of APA is fast convergence with a relative small computational cost, especially for colored signals such as speech. APA updates the filter coefficients by simultaneously minimizing the current and past error signals between the observed desired signals and the filter outputs. In this report, we propose a new adaptive algorithm, the error-memoryless affine projection algorithm (EMAPA), by modifying the second-order APA. Unlike APA, EMAPA does not use the past error signal directly, but uses the approximated past error estimated from the current error signal and the correlation between current and past input signals. By choosing step-size properly, EMAPA can achieve similar characteristics to APA at lower computational cost., The Institute of Electronics, Information and Communication Engineers, 10 Dec. 2004, Technical report of IEICE. EA, 104, 497, 77-82, 0913-5685, 110003284917, AN10164817 - A Method and Efficiency of Transforming Auditory Signals into Vibrations for Perception by Hearing-impaired people
織田修平; 水島昌英; 古家賢一; 羽田陽一; 片岡章俊
21 Oct. 2004, 電子情報通信学会技術研究報告, 104, 388(WIT2004 38-47), 41-46, 0913-5685, 200902240245789613 - 音源方向推定を用いた特定方向収音アダプティブアレー
小林和則; 羽田陽一; 古家賢一; 片岡章俊
21 Sep. 2004, 日本音響学会研究発表会講演論文集, 2004, 629-630, 1340-3168, 200902206152961684 - 聴覚障害者支援を目的とした報知音の振動呈示の有効性について―健聴者による検討―
織田修平; 水島昌英; 古家賢一; 羽田陽一; 片岡章俊
20 Aug. 2004, 情報科学技術フォーラム, FIT 2004, 541-542, 200902218217651910 - Gain Function Emphasis for Echo Reduction in Frequency Domain
SAKAUCHI Sumitaka; HANEDA Yoichi; KATAOKA Akitoshi
Echo reduction based on a short-time spectral amplitude estimation is non-linear processing in the frequency domain. This has a smaller calculation cost than an adaptive filter. It acts robustly when the echo-path changes, and can provide double-talk, which loss insertion, e.g., a voice switch, can not. In this paper, we explain the basic method of calculating the echo reducing gain in the frequency domain, propose an emphasis method for the echo reducing gain that reduces the loss of transmitted speech and improves the echo-suppression level. For characteristics in a frequency bin, the gain emphasis method estimates whether or not the echo or speech is occupied and colors the frequency coefficients of the echo reducing gain. Computer simulation results show that the proposed method improved the echo-suppression level by more than 10 dB without distorting the transmitted speech., The Institute of Electronics, Information and Communication Engineers, 18 Jun. 2004, Technical report of IEICE. EA, 104, 143, 55-60, 0913-5685, 110003283937, AN10164817 - A robust NLMS algorithm for acoustic echo cancellation
SHIMAUCHI Suehiro; HANEDA Yoichi; KATAOKA Akitoshi
Robustness against double-talk is one of the most important issues on adaptive filtering for acoustic echo cancellation. In this report, we propose a new robust algorithm, the gradient-limited NLMS (GL-NLMS), which is derived by modifing the cost function for the NLMS based on M-estimates in robust statistics. The proposed method achieves greater robustness by nonlinearly limiting the gradient used in the NLMS, based on the difference in the distributions of the error-to-reference ratio during single-talk and double-talk., The Institute of Electronics, Information and Communication Engineers, 18 Jun. 2004, Technical report of IEICE. EA, 104, 143, 49-54, 0913-5685, 110003283936, AN10164817 - A-4-48 A Study on Echo Path Imbalance Problem in Stereo Echo Cancellation
Emura Satoru; Haneda Youichi; Kataoka Akitoshi
The Institute of Electronics, Information and Communication Engineers, 08 Mar. 2004, Proceedings of the IEICE General Conference, 2004, 139-139, 110003265778, AN10471452 - The Efficiency of Transforming Auditory Signals into Vibrations for Perception by Hearing-Impaired People-Comparison between Hearing-Impaired People and People with Normal Hearing-
織田修平; 水島昌英; 古家賢一; 羽田陽一; 片岡章俊
2004, ヒューマンインタフェースシンポジウム論文集, 2004 (CD-ROM), 2133, 1345-0794, 200902281570248252 - 騒音下での拡声通話を可能とするノイズ抑圧機能付きエコーキャンセラ技術 (特集 簡単,快適,便利なコミュニケーション環境のための音声・音響処理技術)
阪内 澄宇; 羽田 陽一; 岡本 学
電気通信協会, 2004, NTT技術ジャ-ナル, 16, 1, 14-17, 0915-2318, 80016340414, AN1013019X - Study on frequency domain echo canceller based on higher order statistics
SHIMAUCHI Suehiro; HANEDA Yoichi; KATAOKA Akitoshi
18 Mar. 2003, 日本音響学会研究発表会講演論文集, 2003, 1, 615-616, 1340-3168, 10018036286, AN00351181 - An adaptive algorithm for stereo echo cancellation in noisy environment
EMURA Satoru; HANEDA Youichi
18 Mar. 2002, 日本音響学会研究発表会講演論文集, 2002, 1, 645-646, 1340-3168, 10018034480, AN00351181 - A study for non-linear processing of stereophonic acoustic echo cancellers to stereo signals with a power difference
Nakagawa Akira; Haneda Yoichi
The Institute of Electronics, Information and Communication Engineers, 2002, Proceedings of the IEICE General Conference, 4, 168-168, 110003496359, AN10471452 - A study of adaptive algorithm for stereo signals with a power difference
NAKAGAWA Akira; HANEDA Yoichi
With broadband networks spreading, high-presence acoustic communication like stereophonic full-duplex communication is desired. Thus, a stereophonic acoustic echo canceller is required to avoid howling and echose when loudspeakers and microphones are used for hands- free communication. However, the performance of the acoustic echo canceller is degraded when there is a large power difference between stereo signals received from a remote site; that is the convergence speed of the coefficient error is low. In this paper, we propose normalizing the stereo received signals by the root of the squared summation in the adaptive algorithm of the stereophonic acoustic echo canceller in order to improve the convergence speed. The effectiveness of this is confirmed by computer simulation., The Institute of Electronics, Information and Communication Engineers, 19 Oct. 2001, Technical report of IEICE. EA, 101, 382, 45-52, 0913-5685, 110003284778, AN10164817 - Echo Reduction applied to multi-channel hands-free communication.
SAKAUCHI Sumitaka; TANAKA Masashi; HANEDA Yoichi; YAMAMORI Kazuhiko
01 Oct. 2001, 日本音響学会研究発表会講演論文集, 2001, 2, 539-540, 1340-3168, 10007459354, AN00351181 - Frequency-domain stereo adaptive algorithm enhancing additive signals
EMURA Satoru; HANEDA Youichi
01 Oct. 2001, 日本音響学会研究発表会講演論文集, 2001, 2, 537-538, 1340-3168, 10007459348, AN00351181 - A Study for step size parameter of stereo acoustic echo canceller
Nakagawa Akira; Haneda Yoichi
The Institute of Electronics, Information and Communication Engineers, 07 Mar. 2001, Proceedings of the IEICE General Conference, 2001, 132-132, 110003256951, AN10471452 - A study on adaptive algorithm enhancing decorrelated additive signals for stereo echo canceller
EMURA Satoru; HANEDA Youichi
01 Mar. 2001, 日本音響学会研究発表会講演論文集, 2001, 1, 615-616, 1340-3168, 10007457038, AN00351181 - A desing of a hands-free communication unit using loudspeakers and microphones with a flat directional pattern :
Nakagawa Akira; Haneda Yoichi; Shimauchi Suehiro; Aoki Shigeaki; Makino Shoji
Acoustical Society of Japan, 2001, Acoustical science and technology, 22, 5, 387-387, 110003127945, AA11501808 - 多チャネル音響エコーキャンセラ用適応アルゴリズムの高速化 (特集論文1 エコーキャンセラ技術)
江村 暁; 羽田 陽一; 山森 和彦
電気通信協会, 2001, NTT R & D, 50, 4, 253-258, 0915-2326, 40004828467, AN10076516 - 短時間スペクトラル振幅推定を用いた周囲雑音と残留エコーの抑圧 (特集論文1 エコーキャンセラ技術)
阪内 澄宇; 田中 雅史; 羽田 陽一
電気通信協会, 2001, NTT R & D, 50, 4, 246-252, 0915-2326, 40004828466, AN10076516 - 全指向性を持つスピーカ・マイクロホン一体型通話装置構成法 (特集論文1 エコーキャンセラ技術)
中川 朗; 羽田 陽一; 山森 和彦
電気通信協会, 2001, NTT R & D, 50, 4, 240-245, 0915-2326, 40004828465, AN10076516 - A nonlinear noise and residual echo reduction method with less speech distortion
SAKAUCHI Sumitaka; TANAKA Masashi; HANEDA Yoichi; YAMAMORI Kazuhiko
01 Sep. 2000, 日本音響学会研究発表会講演論文集, 2000, 2, 361-362, 1340-3168, 10005116815, AN00351181 - A multi-channel acoustic echo canceller using channel number compressor and expander
Nakagawa Akira; Shimauchi Suehiro; Haneda Yoichi; Aoki Shigeaki; Makino Shoji
The Institute of Electronics, Information and Communication Engineers, 07 Mar. 2000, Proceedings of the IEICE General Conference, 2000, 140-140, 110003259782, AN10471452 - 家庭に浸透するインターネット端末 (特集 家庭に浸透するマルチメディア通信機器--家庭内へのマルチメディア普及のために)
有海 寿雄; 後藤 勇; 羽田 陽一
電気通信協会, 1999, NTT技術ジャ-ナル, 11, 6, 15-17, 0915-2318, 40004833722, AN1013019X - A study on hands-free unit with a sub-loudspeaker for acoustic echo reduction
Haneda Yoichi; Shimauchi Suehiro; Kaneda Yutaka
The Institute of Electronics, Information and Communication Engineers, 07 Sep. 1998, Proceedings of the Society Conference of IEICE, 1998, 134-134, 110003344747, AN10489017 - Implementation of acoustic echo canceller using duo-filter control on a fixed-point DSP
Shimauchi Suehiro; Haneda Yoichi; Kaneda Yutaka
The Institute of Electronics, Information and Communication Engineers, 07 Sep. 1998, Proceedings of the Society Conference of IEICE, 1998, 243-243, 110003344835, AN10489017 - The combination of the non-linear echo suppression using short-time spectral amplitude estimation and an adaptive filter for acoustic echo cancelling
SAKAUCHI Sumitaka; SASAKI Junko; HANEDA Yoichi
01 Sep. 1998, 日本音響学会研究発表会講演論文集, 1998, 2, 607-608, 1340-3168, 10004299330, AN00351181 - A Study on Configuration of Stereo Echo Canceller with Cross-Correlation Shaker
Shimauchi Suehiro; Haneda Yoichi; Makino Shoji; Kaneda Yutaka
The Institute of Electronics, Information and Communication Engineers, 06 Mar. 1998, Proceedings of the IEICE General Conference, 1998, 121-121, 110003261291, AN10471452 - A study of the non-linear echo suppression using short-time spectral amplitude estimation
SAKAUCHI Sumitaka; HANEDA Yoichi
01 Mar. 1998, 日本音響学会研究発表会講演論文集, 1998, 1, 551-552, 1340-3168, 10002747172, AN00351181 - A study of pre-processing to improve the convergence on a stereo echo canceller
SUZUKI Kuniyasu; SAKAUCHI Sumitaka; SHIMAUCHI Suehiro; HANEDA Yoichi
01 Mar. 1998, 日本音響学会研究発表会講演論文集, 1998, 1, 549-550, 1340-3168, 10002747168, AN00351181 - DSP implementation and evaluation of a complex projection subband acoustic echo canceller that uses two different analysis filters
NAKAGAWA Akira; HANEDA Yoichi
TV会議などの拡声通話系では、通話の妨げとなるエコーやハウリングを消去するために、音響エコーキャンセラが用いられる。 筆者らは、エコーキャンセラの演算量の削減および、収束速度の向上を目的として、フィルタバンク設計などのサブバンド方式の研究や射影アルゴリズムなどの高速適応アルゴリズムの研究を行ってきている。また、高速射影アルゴリズムをサブバンド方式に適用することで、音声入力信号に対し、白色雑音信号とほぼ同様な収束速度が得られることを計算機シミュレーションで確認している。 本稿では、これらの検討結果に基づき、2つの異なる分析フィルタを用いた、複素射影サブバンドエコーキャンセラ(SBEC)をDSPで実現し、収束特性を評価したので報告する。, The Institute of Electronics, Information and Communication Engineers, 13 Aug. 1997, Proceedings of the Society Conference of IEICE, 1997, 92-92, 110003341379, AN10489017 - Study on acoustic coupling in multi-channel teleconference system
SHIMAUCHI Suehiro; HANEDA Yoichi; KOJIMA Junji
01 Mar. 1997, 日本音響学会研究発表会講演論文集, 1997, 1, 631-632, 1340-3168, 10002743556, AN00351181 - Consideration on frequency domain echo return loss required for audio teleconference systems(Part2)
SAKAUCHI Sumitaka; HANEDA Yoichi
01 Mar. 1997, 日本音響学会研究発表会講演論文集, 1997, 1, 623-624, 1340-3168, 10002743536, AN00351181 - 室内伝達開教のモデル化 (<小特集>音場・音響信号のモデルとその分析)
羽田 陽一; 金田 豊
The Acoustical Society of Japan (ASJ), 1997, 日本音響学会誌, 53, 2, 139-146, 0369-4232, 110003136344, AN00186234 - Whitening of the filter coefficient update-vector in the subband echo cancellers
NAKAGAWA Akira; HANEDA Yoichi; MAKINO Shoji
サブバンドエコーキャンセラ(SBEC)では、 間引き率を上げ分割数に近付けると、エリアジングの影響により定常消去量が劣下する。これを避けるために間引き率を下げると、適応フィルタヘの入力信号に帯域通過フィルタの特'性が影響し、収束速度が劣下する。筆者らはこの問題に対し、入力信号と反響信号に異なる特'性の分割フィルタを設定する方法を既に提案した。本報告では、適応フィルタ係数の更新部への入力信号が固定の周波数特性を持つことに注目し、これを固定係数の平坦化フィルタで平坦化することによって収束特性を改善する方法を提案する。, The Institute of Electronics, Information and Communication Engineers, 18 Sep. 1996, Proceedings of the Society Conference of IEICE, 1996, 88-88, 110003336927, AN10489017 - Subband stereo echo canceller using projection algorithm with fast convergence to the true echo path.
MAKINO Shoji; SHIMAUCHI Suehiro; HANEDA Yoichi; NAKAGAWA Akira
01 Sep. 1996, 日本音響学会研究発表会講演論文集, 1996, 2, 549-550, 1340-3168, 10002740792, AN00351181 - A study on prototype filter of subband echo canceller
NAKAAGAWA Akira; HANEDA Yoichi; MAKINO Shoji
サブバンドエコーキャンセラ(SBEC)は、音声の白色化効果による適応フィルタの収束速度向上、間引きによる演算量の低減が望める。その一方で、帯域分割/合成フィルタ処理による遅延や定常消去量の低下が問題となる。本報告では、図1に示すポリフェーズ型SBECの2つの帯域分割用プロトタイプフィルタA(z)、B(z)のフィルタ長および適応フィルタ長に着目し、収束特性の改善方法について検討した。, The Institute of Electronics, Information and Communication Engineers, 05 Sep. 1995, Proceedings of the Society Conference of IEICE, 1995, 75-75, 110003342691, AN10489017 - Study on the echo canceller using the ES Projection algorithm
Makino Shoji; Haneda Yoichi; Tanaka Masashi; Kaneda Yutaka; Kojima Jyunji
エコーキャンセラを実環境で安定に動作させるためには,受話音声の微小音区間に対する対策や,ダブルトーク対策が重要である.ここでは,ES射影アルゴリズムをDuo Filter構成のエコーキャンセラに適用し,速い収束と安定な動作を実現したので報告する., The Institute of Electronics, Information and Communication Engineers, 27 Mar. 1995, Proceedings of the IEICE General Conference, 1995, 349-349, 110003247969, AN10471452 - High Performance Acoustic Echo Cannceller Development
Kojima Junji; Makino Shoji; Haneda Yoichi; Shimauchi Suehiro; Kaneda Yutakat
マルチメディア時代の到来を迎え、遠隔地と通信しているにもかかわらず、あたかも同一の室内にいにような会話ができることや、マイクロホンやスピーカーの位置を意識しないで会話ができるシームレスな音響空間の実現が望まれている。NTTでは、適応アルゴリズムなどの音響信号処理の研究成果を活かし、適応追随性や同時通話時の性能に優れた高性能音響エコーキャンセラを開発したので報告する。, The Institute of Electronics, Information and Communication Engineers, 27 Mar. 1995, Proceedings of the IEICE General Conference, 1995, 348-348, 110003247967, AN10471452 - Study on the echo canceller based on the duo filter system using the ES Projection algorithm
HANEDA Yoichi; MAKINO Shoji; KOJIMA Junji; SHIMAUCHI Suehiro
01 Mar. 1995, 日本音響学会研究発表会講演論文集, 1995, 1, 595-596, 1340-3168, 10002733207, AN00351181 - Duo filter control system for acoustic echo cancellers
Haneda Yoichi; Makino Shoji; Tanaka Masashi; Shimauchi Suehiro; Kojima Junji
音響エコーキャンセラを実環境で動作させるためには、(1)ダブルトーク検出を含め、適応動作制御を如何に行なうか。(2)適応フィルタが音響系を同定していない状態で如何にハウリングを抑えるか。の2点が特に重要となる。(1)のダブルトークの検出技術に関してはこれまで多くの研究がなされてきているが、特に優れた方式はなく、送話信号と受話信号のパワー比較などで行なわれている。また、(2)に関しては音声スイッチとの併用法が提案されているが、音響結合量を予測して最適な挿入損失量を与えないと、結果的に過大な挿入損失を与えてしまい、通話に切断感を与えてしまう。本報告では、ES射影アルゴリズムを用いたDuo Filter Control Systemを提案し、上記2つの問題を解決したので報告する。, The Institute of Electronics, Information and Communication Engineers, 1995, Proceedings of the IEICE General Conference, 350-350, 110003247971, AN10471452 - 室内音場伝達関数の共通極・零モデル化 (シ-ムレスな音響空間の実現を目指して<特集>)
羽田 陽一; 牧野 昭二; 金田 豊
電気通信協会, 1995, NTT R & D, 44, 1, 53-58, 0915-2326, 40004826897, AN10076516
Books and other publications
Lectures, oral presentations, etc.
- FDTD 法を用いた疑似頭の発話放射特性の検討
横田 康介; 羽田 陽一
日本音響学会第151回(2024年春季)研究発表会
08 Mar. 2024 - 仮想スピーカアレイを用いた焦点音源法による仮想音源の奥行き移動幅拡大
末藤 立己; 羽田 陽一
日本音響学会第151回(2024年春季)研究発表会
08 Mar. 2024 - 球面スピーカアレイ内部での 3 次元空間インタラクティブ仮想音源定位
澤尻 晃大; 羽田 陽一
日本音響学会第151回(2024年春季)研究発表会
08 Mar. 2024 - 複数のマイク間相関を用いたバーチャルマイクによる騒音制御の性能評価
吉松 亨真; 信夫 直樹; 伊藤 弘章; 小塚 詩穂里; 鎌土 記良; 羽田 陽一
日本音響学会第151回(2024年春季)研究発表会
06 Mar. 2024 - Time domain virtual sensing method based on a rigid-sphere transfer function for active noise control headrests
N Shinobu; T Yoshimatsu; H Itou; S Kozuka; N Kamado; Y Haneda
Acoustics 2023
07 Dec. 2023
04 Dec. 2023- 08 Dec. 2023 - A virtual sensing method based on plane-wave assumptions for active noise control headrests
T Yoshimatsu; N Shinobu; H Itou; S Kozuka; N Kamado; Y Haneda
Acoustics 2023, Peer-reviewed
05 Dec. 2023
04 Dec. 2023- 08 Dec. 2023 - Context-Awareなニューラルネットワークによる残響付き音声からの擬似RIR生成
宮田 涼司; 羽田 陽一
日本音響学会2023秋季研究発表会
28 Sep. 2023 - 疑似頭スピーカと剛球スピーカの放射特性の比較について
横田 康介; 羽田 陽一
日本音響学会2023秋季研究発表会
26 Sep. 2023 - ANCヘッドレストに向けた剛球モデルに基づく時間領域音圧推定
信夫 直樹; 吉松 亨真; 伊藤 弘章; 小塚 詩穂里; 鎌土 記良; 羽田 陽一
26 Sep. 2023 - 一次経路の純粋遅延削減を用いた信号先読み型バーチャルセンシングANC
吉松 亨真; 信夫 直樹; 伊藤 弘章; 小塚 詩穂里; 鎌土 記良; 羽田 陽一
日本音響学会2023秋季研究発表会
26 Sep. 2023 - 三次元方向に移動可能なインタラクティブ焦点音源の感じ方について
廣橋 美帆; 羽田 陽一
日本音響学会2023秋季研究発表会
26 Sep. 2023 - 頭部近傍マイクと剛球モデルを用いた音圧推定における頭部形状の影響
信夫 直樹; 伊藤 弘章; 鎌土 記良; 小塚 詩穂里; 羽田 陽一
15 Mar. 2023 - 投動作によって投げられたように移動する焦点音源の受聴について
山崎 萌恵; 羽田 陽一
15 Mar. 2023 - 複数誤差マイクの時空間関係を考慮したバーチャルセンシング型能動騒音制御
吉松 亨真; 信夫 直樹; 伊藤 弘章; 鎌土 記良; 小塚 詩穂里; 羽田 陽一
日本音響学会2023春季研究発表会
15 Mar. 2023 - 手の動きに合わせた仮想音源の速度と音量の変化による感じ方の違い
廣橋 美帆; 羽田 陽一
日本音響学会2023春季研究発表会
15 Mar. 2023 - 音の空間共有と空中操作
羽田陽一
2023年電子情報通信学会総合大会
07 Mar. 2023 - Interactive audio and real-time sound field synthesis
羽田陽一
音響学会, 日本音響学会, 北海道科学大学
14 Sep. 2022 - 頭部近傍マイクによる耳元音圧推定手法における騒音到来方向の影響
信夫 直樹; 村田 伸; 伊藤 弘章; 鎌土 記良; 日和﨑 祐介; 羽田 陽一
日本音響学会
14 Sep. 2022 - 頭部包囲型球面マイクロホンアレイによる子音の3次元発話放射特性分析
本地 瑛; 羽田 陽一
日本音響学会
14 Sep. 2022 - 球面スピーカアレイを用いた移動音像レンダリングについて
久保 健輔; 羽田 陽一
日本音響学会
14 Sep. 2022 - 剛球を仮定した頭部近傍複数マイクによる耳元音圧予測型ANCの検討
信夫 直樹; 村田 伸; 伊藤 弘章; 鎌土 記良; 日和﨑 祐介; 羽田 陽一
日本音響学会
11 Mar. 2022 - 正12面体スピーカアレイによる3次元発話放射特性の模擬について
本地 瑛; 任 逸; 羽田 陽一
日本音響学会
10 Mar. 2022 - 手の位置に追従する焦点音源のリアルタイム生成について
廣橋 美帆; 任 逸; 羽田 陽一
日本音響学会
10 Mar. 2022 - 楕円アレイを用いた内側音場再現における受聴領域の伸縮と回転
任 逸; 羽田 陽一
日本音響学会
09 Mar. 2022 - 手の平の位置に追従する焦点音源による定位感と楽しさについて
廣橋美帆; 任 逸; 羽田陽一
立体映像技術研究会（3DIT）
07 Mar. 2022 - 円筒関数の加法定理を用いた2次元外側局所音場再現法
任 逸; 羽田 陽一
日本音響学会, 日本音響学会
07 Sep. 2021 - 音響伝達関数の共通極モデル化
羽田陽一
情報処理学会 音楽情報科学研究会 (SIGMUS), 情報処理学会 音楽情報科学研究会，音声言語情報処理研究会
19 Jun. 2021 - 剛楕円スピーカアレイを用いたビームフォーミングについての検討
任 逸; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - 円筒型スピーカアレイの内部における移動音像再生について
久保 健輔; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - 下方に配置した2つのスピーカによる上方音像の再生
原田 雄典; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - 3次元直方体音場の自動測定および可視化
任 逸; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - DNNを用いた音声からの残響時間及びD/R推定
宮田 涼司; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - 頭部包囲型球面マイクロホンアレイを用いた発話放射特性の分析について
本地 瑛; 羽田 陽一
日本音響学会, 日本音響学会
11 Mar. 2021 - 音響×キャリア
羽田陽一
日本音響学会若手フォーラム企画, 日本音響学会若手フォーラム
09 Mar. 2021 - 剛楕円スピーカアレイを用いた焦点音源生成についての検討
任 逸; 羽田 陽一
日本音響学会, 日本音響学会
10 Sep. 2020 - 2 つの正 12 面体スピーカアレイのビーム反射を用いた奥行き感再生
鎮守 麻穂; 任 逸; 羽田 陽一
日本音響学会
16 Mar. 2020 - 下方に水平配置した平面スピーカアレイによる上方音像の再生
原田 雄典; 羽田 陽一
日本音響学会
16 Mar. 2020 - 円形アレイ収音信号を用いた楕円アレイの内部音場再現
任 逸; 羽田 陽一
日本音響学会
16 Mar. 2020 - 円筒上に配置した複数スピーカによるトランスオーラルシステムの検討
伊東 優生; 羽田 陽一
日本音響学会
16 Mar. 2020 - 楕円スピーカアレイに対する解析的な 2 次元音場再現法
任 逸; 羽田 陽一
２０１９年日本音響学会秋季研究発表会
06 Sep. 2019 - 頭部近傍に配置したスピーカ再生時の音像定位について
草島 慧; 羽田 陽一
２０１９年日本音響学会秋季研究発表会
06 Sep. 2019 - エンドファイア型スピーカアレイを用いた能動騒音制御の検討
秋元 亜門; 羽田 陽一
２０１９年日本音響学会秋季研究発表会
04 Sep. 2019 - スピーカ・マイク一体型球面アレイを用いた反射位置探索
小林 真萌; 羽田 陽一
２０１９年日本音響学会秋季研究発表会
04 Sep. 2019 - 直線スピーカアレイによる多重極音源を用いたビーム幅制御
今泉 健太; 堤 公孝; 中平 篤; 羽田 陽一
日本音響学会２０１９年春季研究発表会
07 Mar. 2019 - 小型ツインスピーカのディジタルイコライジングによる音質補正の検討
合田 梨乃; 羽田 陽一
日本音響学会２０１９年春季研究発表会
07 Mar. 2019 - オフセンター配置スピーカアレイによるトランスオーラル再生の検討
伊東 優生; 羽田 陽一
日本音響学会２０１９年春季研究発表会
07 Mar. 2019 - 円調和展開に基づく 2 つの剛体円形スピーカアレイ用いた外側音場再現
任 逸; 羽田 陽一
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - 正 12 面体マイクロホン・スピーカアレイを用いた反射音探索による室内の壁位置推定
小林 真萌; 羽田 陽一
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - Dropout 変分推論モデルによる多音源方向推定
田中 龍亮; 羽田 陽一
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - 2 個所のステレオ信号からの中間位置ステレオ信号の生成
羽田 陽一
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - 頭部包囲型球面マイクロホンアレイによる声の放射指向特性の測定
秋元 亜門; 伊東 優生; 大林 敬幸; 任 逸; 山里 飛鳥; 羽田 陽一
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - 管路内水中音響データによる漏水音の自動検出―現地流下試験データの検討―
高木 一幸; 石川 佳佑; 羽田 陽一; 浅野 勇; 森 充広; 川上 昭彦; 川邉 翔平
日本音響学会２０１９年春季研究発表会
06 Mar. 2019 - Directivity control using two circular loudspeaker arrays
Y. Ren; Y. Haneda
RISP International Workshop on Nonlinear Circuits, Communications and Signal
06 Mar. 2019 - 球面マイクアレイを用いた両耳信号生成に対する移動感付与
大林敬幸; 羽田陽一
応用音響研究会，信学技報
22 Jan. 2019 - 深層学習モデルを用いた信号区間選択に基づく音源方向推定
田中龍亮; 羽田陽一
応用音響研究会，信学技報
22 Jan. 2019 - Spherical microphone array post-filtering for speech enhancement using PSD estimation in beamspace and the application to unmanned aerial vehicles
R. Kohyama; Y. Haneda
Western Pacific Commission for Acoustics
14 Nov. 2018 - Around-neck loudspeaker array for wearable personal audio reproduction
A. Tada; Y. Haneda
Western Pacific Commission for Acoustics
14 Nov. 2018 - 適応的窓関数切換を利用した MDCT 領域の DNN 音源強調
小泉 悠馬; 原田 登; 羽田 陽一
日本音響学会2018年秋季研究発表会
13 Sep. 2018 - 球面マイクロホンアレイのビームフォーマ出力を利用したポストフィルタの設計とドローンへの応用
上山 了介; 羽田 陽一
日本音響学会2018年秋季研究発表会
12 Sep. 2018 - 管路内水中音響データによる漏水音の自動検出－部分帯域音響モデルによる検討－
高木 一幸; 石川 佳佑; 羽田 陽一; 浅野 勇; 森 充広; 川上 昭彦; 川邉 翔平
日本音響学会2018年秋季研究発表会
12 Sep. 2018 - 2 つの剛体円形スピーカアレイを用いた仮想音源生成
任 逸; 羽田 陽一
日本音響学会2018年秋季研究発表会
12 Sep. 2018 - Temporal DRRと一般化レイリー商に基づく残響環境でのMUSIC
田中龍亮; 羽田陽一
電子情報通信学会，応用音響研究会 信学技報 EA2017-149
19 Mar. 2018 - 焦点音源法で生成した多重極音源の時間領域実装における音場再現精度改善
堤 公孝; 羽田 陽一; 野口 賢一; 高田 英明
日本音響学会2018年春季研究発表会
15 Mar. 2018 - ILD と ITD の特性を考慮した球面マイクロホンアレイからのバイノーラル信号生成手法の検討
大林 敬幸; 羽田 陽一
日本音響学会2018年春季研究発表会
14 Mar. 2018 - 適応ノイズキャンセリングとウィナーフィルタを用いた 2 マイクによるドローン用収音
上山 了介; 羽田 陽一
日本音響学会2018年春季研究発表会
14 Mar. 2018 - 円形スピーカアレイの外側へのモードマッチングによる仮想音源生成
任 逸; 佐藤 航也; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 縦型円形アレイヘッドホンを用いた音像定位の検討
草島 慧; 今泉 健太; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 8ch2 次元 Ambisonics 再生のための収音指向特性の改善
岩附 知宏; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 残響環境におけるMUSIC のための一般化レイリー商二乗誤差最小化基準を用いた部分空間推定法
田中 龍亮; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 深層学習を用いた音源方向推定における方向誤差最小化のための目的関数設計
田中 龍亮; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 修正離散コサイン領域におけるend-to-end DNN 音源強調
小泉 悠馬; 原田 登; 小林 和則; 羽田 陽一
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 管路内水中音響データによる漏水音の自動検出
高木 一幸; 羽田 陽一; 浅野 勇; 森 充広; 川上 昭彦; 川邉 翔平
日本音響学会2018年春季研究発表会
13 Mar. 2018 - 周波数平滑とTemporal DRR に基づくMUSIC 法による残響下での音源方向推定
田中 龍亮; 羽田 陽一
日本音響学会2017年秋季研究発表会
25 Sep. 2017 - 焦点音源で生成した多重極音源による仮想音像の指向性制御
堤 公孝; 高田 英明; 羽田 陽一
日本音響学会2017年秋季研究発表会
25 Sep. 2017 - 円調和・縦型多重極表現に基づく円筒形アレイの3 次元指向性制御
佐藤 航也; 羽田 陽一
日本音響学会2017年秋季研究発表会
25 Sep. 2017 - 広域に分散配置したマイクロホンを連携させる遠方雑音抑圧法の検討
小泉 悠馬; 齊藤 翔一郎; 小林 和則; 島内 末廣; 羽田 陽一
日本音響学会2017年秋季研究発表会
25 Sep. 2017 - 聴感評点を向上させるためのDNN 音源強調関数のブラックボックス最適
小泉 悠馬; 丹羽 健太; 小林 和則; 羽田 陽一
日本音響学会2017年秋季研究発表会
25 Sep. 2017 - 同一方向到来音源に対する遠方音源収音のための2つの指向性制御を用いた非線形処理手法の検討
澤藤圭吾; 羽田陽一
電子情報通信学会 応用音響研究会
21 Jul. 2017 - 直線マイクロホンアレイで収音した信号からの半任意位置両耳信号の模擬
山里 飛鳥; 羽田 陽一
日本音響学会2017年春季研究発表会
15 Mar. 2017 - 広帯域指向性再生を目指した小型エンドファイア型スピーカアレイの実装
多田 明生; 今泉 健太; 佐藤 航也; 羽田 陽一
日本音響学会2017年春季研究発表会
15 Mar. 2017 - 球面マイクロホンアレイを用いた球面調和関数領域での 3 次元バイノーラル信号生成に関する検討
大林 敬幸; 羽田 陽一
日本音響学会2017年春季研究発表会
15 Mar. 2017 - 32 面体スピーカアレイの指向性ビームの反射音を用いた上下感再生
坂本 浩央; 羽田 陽一
日本音響学会2017年春季研究発表会
15 Mar. 2017 - 波数とアレイ信号処理
羽田 陽一
日本音響学会
15 Mar. 2017 - エンドファイアアレイからなる肩掛け型ウェアラブルスピーカによる音像定位
今泉 健太; 羽田 陽一
電子情報通信学会，応用音響研究会
02 Mar. 2017 - 円形アレイを用いた円調和展開モードに基づく3次元指向性制御
佐藤 航也; 羽田 陽一
電子情報通信学会，応用音響研究会
02 Mar. 2017 - 直線マイクロホンアレイ信号からの円筒形両耳信号への変換
山里 飛鳥; 羽田 陽一
電子情報通信学会，応用音響研究会
01 Mar. 2017 - 確率的重み付き最小二乗法に基づく指向性制御
田中 龍亮; 羽田 陽一
電子情報通信学会 応用音響研究会
25 Jan. 2017 - 複数の等方的ビームにおける事前DRR推定を利用した残響抑圧用ポストフィルタ
山本 裕平; 羽田 陽一
Japanese, 電子情報通信学会 応用音響研究会, 電子情報通信学会 応用音響研究会, 静岡県
07 Jul. 2016 - 残響音声の立ち上がり／立ち下がり共分散行列の同時対角化によるMUSIC法
田中 龍亮; 羽田 陽一
Japanese, 電子情報通信学会，応用音響研究会
29 Mar. 2016 - 適応的な重み変更を用いた最小二乗法による正20面体スピーカアレイの指向性制御
大小原 亮; 羽田 陽一
Japanese, 電子情報通信学会，応用音響研究会
29 Mar. 2016 - マイクロホンアレイの開口長疑似拡張を用いた2音源の可視化
Maureen,羽田 陽一
Japanese, 電子情報通信学会，応用音響研究会
29 Mar. 2016 - 目標音圧窓関数と重みの自動化による指向性スピーカアレイフィルタの検討
佐藤 航也; 関 貴志; 羽田 陽一
Poster presentation, Japanese, 電子情報通信学会，応用音響研究会, Domestic conference
29 Mar. 2016 - 音場制御と聴覚マスキングの併用による円形スピーカアレイを用いたエリア再生法の提案
関 貴志; 羽田陽一
Poster presentation, Japanese, 日本音響学会2016年春季研究発表会, Domestic conference
10 Mar. 2016 - 指向性ビームによって作られた音場の直線スピーカアレイ波面合成による再現
今泉 健太; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2016年春季研究発表会, Domestic conference
10 Mar. 2016 - 球面マイクロホンアレイの等方的ビームを利用した残響除去ポストフィルタの設
山本 裕平; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2016年春季研究発表会, Domestic conference
10 Mar. 2016 - 指向性スピーカアレイのサイドローブ抑圧を目的とした目標音圧窓関数の最適化の検討
佐藤 航也; 関 貴志; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2016年春季研究発表会, Domestic conference
10 Mar. 2016 - 発話の立ち上がり／立ち下がり共分散行列を利用した残響下での音源方向推定
田中 龍亮; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2016年春季研究発表会, Domestic conference
10 Mar. 2016 - 正20面体スピーカアレイの指向性制御を用いたトランスオーラルシステムの検討
大小原 亮; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会 応用音響研究会, Domestic conference
03 Jul. 2015 - 4ch ヘッドマウントスピーカによる音像定位手法の検討
嵯峨 俊; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2015年春季研究発表会, Domestic conference
17 Mar. 2015 - FDTD 法を用いたパラボラ反射板ビームフォームミングフィルタの生成方式の検討
中島 仁美; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2015年春季研究発表会, Domestic conference
16 Mar. 2015 - 正12面体スピーカアレイを用いた球面調和関数領域の MV ビームフォーマ
坂東 和奈; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2015年春季研究発表会, Domestic conference
16 Mar. 2015 - 逆伝搬法と焦点アレー法を用いた奥行き方向の音源位置可視化
Maureen,羽田 陽一
Poster presentation, Japanese, 日本音響学会2015年春季研究発表会, Domestic conference
16 Mar. 2015 - 円形スピーカアレーを用いたNBSFC法によるエリア再生法の検討
関 貴志; 羽田陽一
Poster presentation, Japanese, 電子情報通信学会，応用音響研究会, Domestic conference
03 Mar. 2015 - 球面調和関数領域のMVビームフォーマによる正12面体スピーカアレイの指向性制御
坂東和奈; 羽田陽一
Poster presentation, Japanese, 電子情報通信学会，応用音響研究会, Domestic conference
03 Mar. 2015 - FDTD法によるアレイマニフォールドベクトルを用いた反射板ビームフォーミング
中島仁美; 羽田陽一; 丹羽健太
Poster presentation, Japanese, 電子情報通信学会，応用音響研究会, Domestic conference
03 Mar. 2015 - 四重極子スピーカの実測性能について
関 貴志; 羽田陽一
Poster presentation, Japanese, 日本音響学会2014年秋季研究発表会, 日本音響学会, 北海道, Domestic conference
03 Sep. 2014 - 球面マイクロホンアレイに関するアレイマニフォールドベクトルの検討
中島 仁美; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2014年春季研究発表会, 日本音響学会, 東京, Domestic conference
10 Mar. 2014 - 近接平行 2 直線アレイを用いた近傍収音に関する検討
戀川 真己; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2014年春季研究発表会, 日本音響学会, 東京, Domestic conference
10 Mar. 2014 - 正 12 面体スピーカアレイによる球面調和関数の再現について
坂東 和奈; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2014年春季研究発表会, 日本音響学会, 東京, Domestic conference
10 Mar. 2014 - 48ch 円形スピーカアレーによるエバネッセント波生成の検討
関 貴志; 羽田 陽一
Poster presentation, Japanese, 日本音響学会2014年春季研究発表会, 日本音響学会, 東京, Domestic conference
10 Mar. 2014 - エバネッセント波の減衰を模擬した音場制御型エリア再生
伊藤弘章; 古家賢一; 羽田 陽一
Oral presentation, Japanese, 日本音響学会,日本音響学会春季研究発表会
Mar. 2012 - エバネッセント波の減衰を模擬した音場制御型エリア再生
伊藤弘章; 古家賢一; 羽田 陽一
Oral presentation, Japanese, 日本音響学会,日本音響学会春季研究発表会
Mar. 2012 - 残響音の等方到来を仮定した直間比推定
日岡裕輔; 古家賢一; 丹羽健太; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会春季研究発表会
Mar. 2012 - 非線形エコーキャンセラのためのカスケード型LMS アルゴリズムの正規化について
島内末廣; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会春季研究発表会
Mar. 2012 - 直接音と間接音の到来時間差に着目した1 チャネル入力信号中の衝撃音遠近判定
野口賢一; 島内末廣; 大室仲; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会春季研究発表会
Mar. 2012 - スピーカ指向性を考慮した時空間周波数領域での音場再現フィルタ設計手法
小山翔一; 古家賢一; 日和崎祐介; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会春季研究発表会
Mar. 2012 - 音響イベントのクラスタ分類に基づく行動識別手法の検討
井本桂右; 島内末廣; 大室仲; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会春季研究発表会
Mar. 2012 - 大型多重反射板付きマイクロホンアレーを用いた超指向性収音
丹羽 健太; 阪内 澄宇; 古家 賢一; 岡本 学; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会秋季研究発表会
Sep. 2011 - 円形スピーカアレーを用いたエバネッセント波再生手法について
伊藤 弘章; 古家 賢一; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会秋季研究発表会
Sep. 2011 - 音場収音・再現のための時空間周波数領域信号変換法
小山 翔一; 古家 賢一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会秋季研究発表会
Sep. 2011 - 時空間スペクトルの位相シフトによる音場再現位置の制御
小山 翔一; 古家 賢一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会秋季研究発表会
Sep. 2011 - 反射率可変の壁面音響システムを用いた音場再生システム
岩永 景一郎; 尾本 章; 河原 一彦; 古家 賢一; 清原 健司; 岡本 学; 羽田 陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集,日本音響学会秋季研究発表会
Sep. 2011 - コミュニケーショントリガ：音環境状態遷移に基づくシームレスコミュニケーション
野口賢一; 阪内澄宇; 島内末廣; 大室仲; 羽田陽一
Oral presentation, Japanese, 画像電子学会年次大会
Jun. 2011 - 狭指向性を実現する反射板利用型マイクロホンアレー
丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - 角度スペクトル微分を用いた波面合成法の実音場評価
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - 球調和関数展開に基づく多重極音源を用いた任意の指向性制御
古家賢一; 羽田陽一; 尾本章; 河原一彦
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - 球調和関数展開に基づく多重極音源型エンドファイアスピーカアレー
羽田陽一; 古家賢一; 伊藤弘章; 尾本章; 河原一彦
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - 多様な室内残響下における音源放射指向特性と残響成分の分析
岡本拓磨; 丹羽健太; 阪内澄宇; 羽田陽一; 岩谷幸雄; 鈴木陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - 直線状スピーカアレーを用いたエバネッセント波再生手法について
伊藤弘章; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2011 - エリア再生のための多点制御法による音波の減衰制御
伊藤弘章; 古家賢一; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Feb. 2011 - ITU-T G.722/G.711.1SWB標準化候補におけるモード選択を用いた8-14kHz帯域符号化手法
福井勝宏; 栗原祥子; 佐々木茂明; 日和崎祐介; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 角度スペクトル微分による音圧勾配取得に基づく波面合成法
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 伝達経路の時間応答特性を考慮した相関関数推定法と残響除去への適用
江村暁; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - エバネッセント波を用いたエリア再生に関する研究
伊藤弘章; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 最小分散ビームフォーマを用いた空間周波数解析による反射音情報推定
丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 誤差信号の統計モデルを考慮したステレオエコーキャンセラ
島内末廣; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 収音･再生系におけるAD-DA間のサンプリング周期ずれに対応した音響エコーキャンセラ
齊藤翔一郎; 中川朗; 島内末廣; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 逆伝播演算による波面合成位置の操作
小山翔一; 日和崎祐介; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2010 - 多段エコー推定による多チャネルエコー消去法
江村暁; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - 二次元格子状マイクロホンアレーを用いた反射音情報の推定
丹羽健太; 日岡裕輔; 阪内澄宇; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - 線状スピーカアレーを用いたエリア再生について
伊藤弘章; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - 時間領域差分法による音場解析における反射境界条件の設定について
木下郁一郎; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - 多地点会議のためのステレオ音声レンダリング-スケーラブル音声符号化の低演算量部分選択ミキシング-
堤公孝; 福井勝宏; 栗原祥子; 佐々木茂明; 日和崎祐介; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - 高臨場遠隔通信のためのサウンドウォールシステム
阪内澄宇; 古家賢一; 小山翔一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2010 - ITU-T G.722/G.711.1 14kHz帯域音声符号化標準化候補アルゴリズムの性能について
堤公孝; 福井勝宏; 栗原祥子; 佐々木茂明; 日和崎祐介; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2009 - 音源の特性に応じたモード選択に基づく8-14kHz帯域符号化手法-A ITU-T G.722/G.711.1Super Wideband Extension標準化候補-
福井勝宏; 堤公孝; 佐々木茂明; 日和崎祐介; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2009 - 受音信号の直間比に基づく距離別収音の検討
日岡裕輔; 丹羽健太; 阪内澄宇; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2009 - 音源向きと空間相関行列の固有空間の関連性
丹羽健太; 日岡裕輔; 阪内澄宇; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2009 - 参照信号ベクトルに応じたフィルタ係数群を持つ非線形歪対応エコーキャンセラ
齊藤翔一郎; 福井勝宏; 中川朗; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2009 - 音声帯域と単語了解度の関係について
栗原祥子; 日和崎祐介; 佐々木茂明; 岡本学; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2009 - 空間相関行列の固有値の比率に着目した発話者向きの推定
丹羽健太; 阪内澄宇; 岡本学; 古家賢一; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2009 - 音響エコー抑圧のための高精度エコーパワー推定
福井勝宏; 中川朗; 島内末廣; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2009 - 残響と雑音が存在する環境下での強調処理音声の品質評価
古家賢一; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2008 - 自由空間伝達関数を用いた多点制御に基づく小型スピーカアレーの実空間性能
羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2008 - 入出力相互相関に着目した音声歪みを軽減するエコー抑圧ゲイン推定方式
福井勝宏; 中川朗; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2008 - 入力信号の時間変動性に着目した高精度雑音推定
福井勝宏; 島内末廣; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2007 - 正規化誤差の統計モデルに基づく音響エコー消去アルゴリズムについて
島内末廣; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2007 - 時間-周波数領域に着目した音響結合量推定方式の評価
福井勝宏; 島内末廣; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2007 - スピーカの非線形性に対応可能なマイクロホン対を用いた音響エコーキャンセラ
小林和則; 古家賢一; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2006 - 方向別AGC機能のIP電話会議装置への実装
小林和則; 羽田陽一; 日和崎祐介; 大室仲; 入島勉; 中山圭一; 阿部匡伸
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2006 - 広帯域音声コーデック(UEMCLIP)のIP電話会議装置への実装
日和崎祐介; 大室仲; 小林和則; 羽田陽一; 入島勉; 中山圭一; 阿部匡伸
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2006 - センタースピーカの先行音効果を用いたステレオ音声の受聴位置拡大
植松尚; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2005 - マイクロホンと音源位置の同時推定-1つ以上のマイクロホン間距離が既知の場合-
小林和則; 古家賢一; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2005 - 焦点形成スピーカアレーに対する空間窓の効果
羽田陽一; 植松尚; 片岡章俊; 佐野綾子
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2005 - 調波構造の強調と帯域別抑圧を組合せた1チャネル突発性雑音抑圧
野口賢一; 阪内澄宇; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2005 - 付加信号強調型ステレオ適応アルゴリズムの収束特性
江村暁; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2005 - 周波数領域における送話信号選択を用いた多チャンネルエコー抑圧処理に関する検討
福井勝宏; 島内末広; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2005 - 高臨場感通信に求められるエコーキャンセラとその課題-ステレオから多チャンネルへ-
島内末広; 江村暁; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2005 - 音源方向推定を用いた特定方向収音アダプティブアレー
小林和則; 羽田陽一; 古家賢一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2004 - スピーカアレーを用いた焦点虚音像の連続移動について
羽田陽一; 植松尚; 片岡章俊; 守谷健弘; 佐野綾子
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2004 - 音場の局所化におけるスピーカ･マイクロホン配置について
植松尚; 羽田陽一; 片岡章俊; FIGGEN S
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2004 - 聴覚障害者支援を目的とした報知音の振動呈示の有効性について-健聴者による検討-
織田修平; 古家賢一; 羽田陽一; 片岡章俊; 水島昌英
Oral presentation, Japanese, 情報科学技術フォーラム
Aug. 2004 - 聴覚障害者支援を目的とした報知音の振動呈示の有効性-聴覚障害者と健聴者の比較-
織田修平; 古家賢一; 羽田陽一; 片岡章俊; 水島昌英
Oral presentation, Japanese, ヒューマンインタフェースシンポジウム論文集
Aug. 2004 - 重み付け多点制御に基づく音場の局所化について
植松尚; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2004 - 多点制御に基づく指向性再生の検討について
羽田陽一; 植松尚; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2004 - 1チャネル入力信号中の突発性雑音の判別と除去
野口賢一; 阪内澄宇; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2004 - ステレオエコーキャンセラのEcho Path Imbalance問題について
江村暁; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2004 - 聴覚障害者への振動呈示方法に関する基礎検討
織田修平; 羽田陽一; 片岡章俊; 水島昌英
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2004 - 高次統計量を用いた周波数領域エコーキャンセラの検討
島内末広; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2003 - 音の到来方向に対する頭部伝達関数の留数変化について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2003 - 1チャネル音声強調処理の低遅延化の検討
阪内澄宇; 羽田陽一; 片岡章俊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2002 - 雑音環境下でのステレオエコー消去用適応アルゴリズム
江村暁; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2002 - パワー比の異なるステレオ信号へのステレオエコーキャンセラ非線形前処理に関する一検討
中川朗; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2002 - 付加信号強調型の周波数領域ステレオ適応アルゴリズム
江村暁; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Oct. 2001 - 多チャネル拡声系へのエコーリダクションの適用
阪内澄宇; 田中雅史; 羽田陽一; 山森和彦
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Oct. 2001 - 共通極･留数モデルに基づく仮想マイクロホンを用いた能動騒音制御について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Oct. 2001 - ステレオエコーキャンセラのための付加信号を強調する適応アルゴリズムについて
江村暁; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 2001 - ステレオエコーキャンセラのステップサイズに関する一検
中川朗; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2001 - 音声歪みを軽減した雑音･エコー非線形除去方式
阪内澄宇; 田中雅史; 羽田陽一; 山森和彦
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 2000 - チャネル数変換型多チャネル音響エコーキャンセラ
中川朗; 島内末広; 羽田陽一; 青木茂明; 牧野昭二
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 2000 - 音響結合低減用サブスピーカを持つ小型拡声通話装置の検討
羽田陽一; 島内末広; 金田豊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Sep. 1998 - デュオフィルタ方式音響エコーキャンセラの固定小数点DSPによる実現
島内末広; 羽田陽一; 金田豊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Sep. 1998 - マスキング効果を用いた低歪み雑音低減方式における効果的な原音付加率の検討
佐々木潤子; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1998 - STSA推定に基づいた非線形エコー抑圧と適応フィルタの組合せ
阪内澄宇; 佐々木潤子; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1998 - 相互相関の変動付加処理に適したステレオエコーキャンセラの構成の検討
島内末広; 羽田陽一; 牧野昭二; 金田豊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1998 - 短時間スペクトラル振幅推定に基づいた非線形エコー抑圧処理の検討
阪内澄宇; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1998 - ステレオエコーキャンセラにおける収束改善のための前処理方式の検討
鈴木邦和; 阪内澄宇; 島内末広; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1998 - 2つの異なる分析フィルタを用いた複素射影サブバンドエコーキャンセラのDSP実現と収束特性評価
中川朗; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Sep. 1997 - 共通極･留数モデルを用いた室内伝達関数の予測について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1997 - マスキング効果を考慮した低歪み一入力系雑音低減方式の検討
佐々木潤子; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1997 - 拡声通信システムにおける周波数帯域別所要エコー抑圧量の検討(その2)
阪内澄宇; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1997 - マルチチャネル拡声通話系の音響結合量について
島内末広; 羽田陽一; 小島順治
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1997 - サブバンドエコーキャンセラにおけるフィルタ係数更新ベクトルの平坦化の検討
中川朗; 羽田陽一; 牧野昭二
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Sep. 1996 - 共通極･留数モデルを用いた室内伝達関数の推定について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1996 - 射影アルゴリズムを用いたサブバンドステレオエコーキャンセラ
牧野昭二; 島内末広; 羽田陽一; 中川朗
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1996 - 室内伝達関数の共通極とその留数の変化について
羽田陽一
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1996 - 損失制御を用いた帯域分割型雑音低減方式について
佐々木潤子; 羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1996 - 室内伝達関数の全極モデルに着目した残響抑圧について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1996 - サブバンドエコーキャンセラのプロトタイプフィルタの検討
中川朗; 羽田陽一; 牧野昭二
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Sep. 1995 - 複素射影サブバンドエコーキャンセラに関する検討
中川朗; 羽田陽一; 牧野昭二
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1995 - エコーキャンセラ用SSBサブバンド射影アルゴリズム
牧野昭二; 羽田陽一; 中川朗
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Sep. 1995 - ES射影アルゴリズムを用いたデュオフィルタ構成のエコーキャンセラの検討
羽田陽一; 牧野昭二; 小島順治; 島内末広
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1995 - 高性能音響エコーキャンセラの開発
小島順治; 牧野昭二; 羽田陽一; 島内末広; 金田豊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1995 - ES射影アルゴリズムの音響エコーキャンセラへの適用
牧野昭二; 羽田陽一; 田中雅史; 島内末広; 小島順治
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1995 - 音響エコーキャンセラ用デュオフィルタコントロールシステム
羽田陽一; 牧野昭二; 田中雅史; 島内末広; 小島順治
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1995 - 頭部伝達関数の方向別零点について
羽田陽一
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Jan. 1994 - 共通極を用いた多点イコライゼーションフィルタについて
羽田陽一; 牧野昭二
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1993 - 複数の室内音場伝達関数に共通な極の最小2乗推定について
羽田陽一; 牧野昭二; 金田豊
Oral presentation, Japanese, 電子情報通信学会大会講演論文集
Mar. 1993 - 共通極を用いたスピーカ特性の多点イコライゼーションについて
羽田陽一; 牧野昭二
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Jan. 1993 - 音響エコーキャンセラにおけるダブルトーク制御方式の検討
中原宏之; 吉川昭吉郎; 羽田陽一; 牧野昭二
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1992 - 室内音場伝達関数の極の推定について
羽田陽一; 牧野昭二; 金田豊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Mar. 1991 - 音の到来方向によらない頭部伝達関数の共通極とモデル化について
羽田陽一; 牧野昭二; 金田豊
Oral presentation, Japanese, 日本音響学会研究発表会講演論文集
Jan. 1991 - 帯域分割形指数重み付けアルゴリズムを用いた音響エコーキャンセラ
牧野昭二; 羽田陽一
Oral presentation, Japanese, 電子情報通信学会全国大会講演論文集
Sep. 1990
Courses
- 知能機械工学実験
The University of Electro-Communications - 知能機械工学実験
電気通信大学 - マルチメディア処理（Ⅱ類）
The University of Electro-Communications - マルチメディア処理（Ⅱ類）
電気通信大学 - Graduate Technical English
The University of Electro-Communications - 大学院技術英語
電気通信大学 - 情報メディアシステム
The University of Electro-Communications - 音声音響情報処理
The University of Electro-Communications - K課程輪講
The University of Electro-Communications - K課程輪講
The University of Electro-Communications - 音声音響情報処理
The University of Electro-Communications - 音声音響情報処理
電気通信大学 - 総合コミュニケーション科学
The University of Electro-Communications - 総合コミュニケーション科学
電気通信大学 - 技術英語
The University of Electro-Communications - 技術英語
電気通信大学 - 総合情報学専攻基礎
The University of Electro-Communications - 総合情報学専攻基礎
電気通信大学 - コンピュータリテラシー
The University of Electro-Communications - 総合情報学基礎
The University of Electro-Communications - 総合情報学基礎
電気通信大学 - 情報メディアシステム
The University of Electro-Communications - 情報メディアシステム
電気通信大学 - Audio signal processing
The University of Electro-Communications - 音響信号処理
電気通信大学 - コンピュータリテラシー
The University of Electro-Communications - コンピュータリテラシー
電気通信大学
Research Themes
- 3D発話放射特性の空間再現と臨場感
羽田 陽一
日本学術振興会, 科学研究費助成事業, 電気通信大学, 基盤研究(C), 22K12071
Apr. 2022 - Mar. 2025 - 3D spatial control of speech for next generation conversation support
01 Apr. 2019 - 31 Mar. 2021 - ウェアラブルスピーカアレーの実現に向けた聴感的指向特性生成技術
Haneda Yoichi
Japan Society for the Promotion of Science, Grants-in-Aid for Scientific Research, The University of Electro-Communications, Grant-in-Aid for Scientific Research (C), We have investigated the wearable loudspeaker array which has multiple loudspeaker units. It can control directivity pattern. In this study, we considered the circular, cylindrical, and linear loudspeaker arrays. When the loudspeaker units were placed on the circle with equal space, we can use the spatial Fourier transform. We can calculate filter coefficients for sound field or directivity control analytically. However, the performance of three dimensional directivity pattern made by two dimensional circular array had not been studied so far. Moreover, if the two cylindrical arrays are located closely, the multiple scattering occurs, but the performance of this situation also has not been investigated. Therefore, we investigated three dimensional directivity pattern with those conditions analytically. Furthermore, we studied the wearable loudspeaker with two linear loudspeaker arrays which indicates the independent sounds at both ears, and prevent the sound leakage around the uses., 16K00232
Apr. 2016 - Mar. 2019 - 腕輪型ハンズフリーフォン向け指向性スピーカアレーの研究
（一財）テレコム先端技術研究支援センター, 奨学寄附金
Jun. 2016 - Mar. 2018 - 3次元音響空間ウォークスルー実現に向けた波数領域収音再生技術
KANEDA Yutaka; NISHIJIMA Keisuke
Japan Society for the Promotion of Science, Grants-in-Aid for Scientific Research, Tokyo Denki University, Grant-in-Aid for Scientific Research (B), Auditory walkthrough system, which analyzes and synthesizes the main sound source information (source signal and its location) and the ambient information (room impulse responses, reverberations, noises) from sound field, is proposed. We proposed the fast and high accuracy measurement methods of multiple impulse responses and the source separation method based on the difference between the spatial spectrum information. Furthermore, we made a prototype system in which the user can walk through virtually in the restricted reproduced-sound-field., 15H02728
Apr. 2015 - Mar. 2018 - 球面スピーカアレーによる高精細音響再生に関する研究
(公財)電気通信普及財団, 研究助成
01 Apr. 2014 - 31 Mar. 2016 - 音の距離を識別する３Ｄマイクロホンアレー
公益財団法人栢森情報科学振興財団
22 Jan. 2014 - 15 Nov. 2015 - 狭指向性と全方位性を両立する指向性スピーカアレーの研究
Scientific research (C), general, Principal investigator
2013 - 2015 - 空間を超越するインタラクティブ聴覚拡張システムの研究
Scientific research (B), general, Coinvestigator
2013 - 2015
Industrial Property Rights
- 音圧推定装置、音圧推定方法、プログラム
Patent right, 伊藤 弘章, 小塚 詩穂里, 鎌土 記良, 信夫 直樹, 吉松 亨真, 羽田 陽一, 特願2023-144105, Date applied: 06 Sep. 2023 - 音圧予測システム、音圧予測装置、音圧予測方法、プログラム
Patent right, 伊藤 弘章, 小塚 詩穂里, 鎌土 記良, 吉松 亨真, 信夫 直樹, 羽田 陽一, 特願2023-144127, Date applied: 06 Sep. 2023 - 生成装置、生成方法、およびプログラム
Patent right, 伊藤 弘章, 鎌土 記良, 小塚 詩穂里, 吉松 亨真, 羽田 陽一, 特願2023-023936, Date applied: 20 Feb. 2023 - 生成装置、生成方法、およびプログラム
Patent right, 伊藤 弘章, 鎌土 記良, 小塚 詩穂里, 信夫 直樹, 羽田 陽一, 特願2023-023937, Date applied: 20 Feb. 2023 - 生成装置、生成方法、およびプログラム
Patent right, 伊藤, 弘章, 村田, 伸, 鎌土, 記良, 信夫 直樹, 羽田 陽一, 特願2022-025303, Date applied: 22 Feb. 2022 - フィルタ係数決定装置、フィルタ係数決定方法、プログラム、および音響システム
Patent right, 羽田陽一, 佐藤航也, 特願2017-173419, Date applied: 08 Sep. 2017 - フィルタ係数決定装置、フィルタ係数決定方法、プログラム、および再生システム
Patent right, 羽田 陽一, 大小原 亮, 佐藤 航也, 関 貴志, 特願2016-032282, Date applied: 23 Feb. 2016 - 音源方向推定装置、音源方向推定方法、およびプログラム
Patent right, 2016-032281, Date applied: 23 Feb. 2016 - 収音装置および収音方法,並びにプログラム
Patent right, 羽田陽一, 戀川真己, 特願2014-39642, Date applied: 28 Feb. 2014
Social Contribution Activities
- 大学見学対応（厚木高校）
Lecturer, Others
20 Oct. 2023 - 日本音響学会第20回サマーセミナー「音響学の基礎と最近のトピックス」
Appearance, 一般社団法人 日本音響学会, 空間フーリエ変換の基礎と応用, 白馬, 音響学を志す若手研究者向けのセミナーに講師として参加。
03 Sep. 2018 - 日本音響学会第19 回サマーセミナー「音響学の基礎と最近のトピックス」
Appearance, 一般社団法人 日本音響学会, 伝達系のモデル化- ""極と零点"", 白馬, 音響学を志す若手研究者向けのセミナーに講師として参加。
10 Sep. 2017 - 夢ナビ ライブ２０１５
Appearance, 株式会社フロムページ, 音を空間で操る, 東京, 高校生 高校生に大学での理工系における研究活動，基礎学問の重要性を理解してもらうことを目的に講義を行った。
11 Jul. 2015
Academic Contribution Activities
- 日本音響学会東海支部70周年祈念行事 来賓挨拶
Competition etc, Others, 日本音響学会東海支部, 25 Sep. 2023 - Inter Noise 2023 来賓挨拶
Competition etc, Others, Inter Noise 2023実行委員会, 20 Aug. 2023 - 日本音響学会2019年春季研究発表会実行委員会委員長
Academic society etc, Planning etc, 日本音響学会, 05 Mar. 2019, 電通大で開催させていただきまして大変ありがとうございました。 発表件数５５０件程度，参加者数１４００名と近年にない盛況ぶりでした。