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 Conference Papers (Available on Advance Programs)  (Sort by: Date Descending)
 Results 1 - 14 of 14  /   
Committee Date Time Place Paper Title / Authors Abstract Paper #
SP, EA, SIP 2020-03-03
09:00
Okinawa Okinawa Industry Support Center
(Cancelled but technical report was issued)
[Poster Presentation] Study on method for calculating loudness of stationary sound using auditory filterbank
Takuto Isoyama, Shunsuke Kidani, Masashi Unoki (JAIST) EA2019-148 SIP2019-150 SP2019-97
A time-domain loudness model is necessary to calculate sound quality metrics such as roughness and fluctuation strength.... [more] EA2019-148 SIP2019-150 SP2019-97
pp.273-278
EA, SIP, SP 2019-03-15
15:15
Nagasaki i+Land nagasaki (Nagasaki-shi) A Basic Study on Azimuth Estimation Model of Sound Image in Car Cabin by Using a Gammachirp Auditory Filterbank
Koji Sakamoto (Denso Ten Technology), Masashi Nakamura, Masanobu Maeda (Denso Ten) EA2018-159 SIP2018-165 SP2018-121
Accuracy verification of azimuth estimation of sound image was performed by applying the frequency domain binaural model... [more] EA2018-159 SIP2018-165 SP2018-121
pp.353-358
WIT, SP 2018-10-27
16:50
Fukuoka Kyushu Institute of Technology(Kitakyushu) Implementation of hearing impairment simulator based on the gammachirp auditory filterbank and its educational application
Toshie Matsui (TUT), Hideki Banno (Meijo Univ.), Ryuichi Nisimura, Toshio Irino (Wakayama Univ.) SP2018-38 WIT2018-26
We have developed and improved a hearing impairment (HI) simulator that allows normal hearing people to experience diffi... [more] SP2018-38 WIT2018-26
pp.31-36
EA, ASJ-H 2017-10-21
14:30
Toyama Ushidake-Onsen Study on modulation spectrum analysis for speech and non-speech signals
Takuto Isoyama, Masashi Unoki (JAIST) EA2017-49
This paper aims to clarify available features for discriminating and classifying speech and non-speech signals.Modulatio... [more] EA2017-49
pp.29-34
SITE, EMM, ISEC, ICSS, IPSJ-CSEC, IPSJ-SPT [detail] 2017-07-15
13:25
Tokyo   Investigation of spikegram-based signal representation for speech fingerprints
Dung Kim Tran, Masashi Unoki (JAIST) ISEC2017-32 SITE2017-24 ICSS2017-31 EMM2017-35
This paper investigates the ability of spikegrams in representing the speech content and voice identications of speech s... [more] ISEC2017-32 SITE2017-24 ICSS2017-31 EMM2017-35
pp.241-246
EA, ASJ-H 2015-08-03
13:30
Miyagi Tohoku Univ., Research Inst. of Electrical Communication Study on restoration of instantaneous amplitude and phase of speech signal in noisy reverberant environments
Yang Liu, Naushin Nower (JAIST), Yonghong Yan (IOA), Masashi Unoki (JAIST) EA2015-11
We have proved that restoring the instantaneous amplitude as well as instantaneous phase on Gammatone filterbank plays a... [more] EA2015-11
pp.7-12
AP, RCS
(Joint)
2014-11-12
13:10
Yamagata Yamagata University, Yonezawa Campus Running-approximation of input-operators of an operator-filterbank by output-operators in a bounded linear-operator-space
Yuichi Kida (Ohu Univ.), Takuro Kida (Tokyo Inst. of Tech.) RCS2014-199
In this paper, based on a fact that a compact Banach-space and a compact Hilbert-space can be covered by a finite $epsil... [more] RCS2014-199
pp.13-18
SP, WIT, ASJ-H 2014-06-19
16:00
Ishikawa   Instantaneous amplitude and phase restoration using Kalman filter for speech enhancement
Yang Liu, Naushin Nower, Masashi Unoki (JAIST) SP2014-51 WIT2014-6
This paper proposes a method of restoring the instantaneous amplitudes and phases using Kalman filter with linear predic... [more] SP2014-51 WIT2014-6
pp.27-32
SP, EA, SIP 2013-05-17
11:10
Okayama   A Note on Fast Computation Method of Polyphase Filterbank
Gen Fujishiro, Koichi Ichige (Yokohama National Univ.) EA2013-18 SIP2013-18 SP2013-18
This paper presents a way of fast computation method in polyphase filterbanks.
Filterbanks divide a single-channel sig... [more]
EA2013-18 SIP2013-18 SP2013-18
pp.103-108
SP, IPSJ-SLP 2012-12-21
14:40
Tokyo TITECH(Ookayama) Reduction of cross spectrum for feature-domain sound source separation
Atsushi Ando (Nagoya Univ.), Kenta Niwa (NTT), Norihide Kitaoka, Kazuya Takeda (Nagoya Univ.) SP2012-93
Speech source separation is utilized for recognition of simultaneous speech. Conventional source separation methods, esp... [more] SP2012-93
pp.107-112
SP 2011-07-22
09:00
Hokkaido Jozankei Grand Hotel Estimation of vocal tract length ratio using auditory filterbank
Erika Okamoto, Toshio Irino, Ryuichi Nisimura, Hideki Kawahara (Wakayama Univ.) SP2011-43
Vocal tract length normalization(VTLN) is an important issue in speech applications, such as automatic speech recognitio... [more] SP2011-43
pp.11-16
SAT 2008-11-06
16:40
Overseas Riviera Hotel Pusan A Prototype Modem for Hyper-Multipoint Data Gathering SATCOM Systems -- A Group Modem Applicable to Arbitrarily and Dynamically Assigned FDMA Signals --
Kiyoshi Kobayashi, Fumihiro Yamashita, Jun-ichi Abe (NTT) SAT2008-30
This paper presents a prototype group modem for a hyper-multipoint data gathering satellite communication system. It can... [more] SAT2008-30
pp.81-86
RCS, SIP
(Joint)
2006-01-26
11:20
Fukuoka Kyusyu Univ. A filterbank based equalization technique for OFDM systems with insufficient guard interval
Seiichiro Horikawa, Hideo Kasami, Makoto Tsuruta, Hidehiro Matsuoka, Hiroki Shoki (Toshiba)
OFDM (Orthogonal Frequency Division Multiplexing)systems operated in the Single Frequency Network (SFN) might face the p... [more] RCS2005-128
pp.25-30
SIP 2005-01-21
15:15
Aichi Nagoya Institute of Technology Interpolation by Multiple Generators and Its Applications
Koichi Ichige (Yokohama National Univ.)
Standard linear interpolation is still worth due to its low computational complexity, even though the inherent problem o... [more] SIP2004-122
pp.73-78
 Results 1 - 14 of 14  /   
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