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 Conference Papers (Available on Advance Programs)  (Sort by: Date Descending)
 Results 1 - 20 of 24  /  [Next]  
Committee Date Time Place Paper Title / Authors Abstract Paper #
CAS, CS 2024-03-14
13:30
Okinawa   Characterization of Semantic Communications in Speech Signal Transmission
Futo Iwanaga, Daisuke Umehara (Kyoto Inst. of Tech.) CAS2023-118 CS2023-111
In recent years, the volume of data in data communication has surged, Characterization of Semantic Communications in Spe... [more] CAS2023-118 CS2023-111
pp.41-46
SP, IPSJ-SLP, EA, SIP [detail] 2023-03-01
14:40
Okinawa
(Primary: On-site, Secondary: Online)
Anomalous sound detection based on differential features of multi channel acoustic signals considering spatial and temporal variations
Shota Nishiyama, Akira Tamamori (AIT)
Anomalous sound detection plays an essential role in machine condition management in factory automation. The task of ano... [more]
CAS, ICTSSL 2021-01-29
15:45
Online Online Reproduction of Japanese drumming rhythm by Deep Neural Network(DNN)
Kazumi Okamoto, Hiroshi Tamura (Chuo Univ.) CAS2020-69 ICTSSL2020-54
Recently, artificial intelligence has been applied to music, such as automatic music generation. In this paper, we propo... [more] CAS2020-69 ICTSSL2020-54
pp.158-161
EA 2020-12-14
10:05
Online Online Speech Signal Detection Based on Bayesian Estimation by Observing Air-Conducted Speech under Existence of Surrounding Noise with Aid of Bone-Conducted Speech
Akira Ikuta, Hisako Orimoto (Prefectural Univ. of Hiroshima), Kouji Hasegawa (Hiroshima Prefectural Technology Research Inst.) EA2020-48
When applying speech recognition systems to actual circumstances such as inspection and maintenance operations in indust... [more] EA2020-48
pp.13-18
ISEC, SITE, ICSS, EMM, HWS, BioX, IPSJ-CSEC, IPSJ-SPT [detail] 2019-07-24
12:10
Kochi Kochi University of Technology Recording device identification based on audio distortion depending on system-on-chip
Akira Nishimura (Tokyo Univ. Info. Sci.) ISEC2019-48 SITE2019-42 BioX2019-40 HWS2019-43 ICSS2019-46 EMM2019-51
This study addresses device-specific distortion observed in recorded
audio, to identify a built-in system-on-a-chip (... [more]
ISEC2019-48 SITE2019-42 BioX2019-40 HWS2019-43 ICSS2019-46 EMM2019-51
pp.311-316
US, EA
(Joint)
2016-01-29
10:00
Osaka Kansai University, Centenary Memorial Hall Multiple devices' synchronization and positional estimation for auditory localization in a confined space
Yosuke Ino, Yuki Ishikawa, Yusuke Naka, Tomoko Yonezawa (Kansai Univ.) EA2015-58
In this paper, we propose an auditory localization system with multiple smart devices by distributing appropriate sound ... [more] EA2015-58
pp.19-26
WIT, SP, ASJ-H, PRMU 2015-06-18
14:25
Niigata   Non-Audible Murmur Enhancement Method using Air- and Body-Conductive Microphones in Noisy Environments and its Evaluation
Yusuke Tajiri, Kou Tanaka, Tomoki Toda, Graham Neubig, Sakriani Sakti, Satoshi Nakamura (NAIST) PRMU2015-42 SP2015-11 WIT2015-11
As one of the silent speech interfaces, Non-Audible Murmur (NAM) microphone which can detect an extremely soft whispered... [more] PRMU2015-42 SP2015-11 WIT2015-11
pp.59-64
EA 2014-06-27
13:30
Mie Mie Univ. An experimental study of optimum parameters in hybrid modulation based on amplitude and frequency with parametric loudspeaker for suitable speech reproduction
Toru Iwasaki, Daisuke Ikefuji, Masato Nakayama, Takanobu Nishiura (Ritsumeikan Univ.) EA2014-6
Parametric loudspeaker with sharper directivity has been generally used for audio guidance to a specific listener. To em... [more] EA2014-6
pp.7-12
NC, MBE
(Joint)
2014-03-17
10:40
Tokyo Tamagawa University Correlation between flicker value and voice signals to tiredness
Ryota Inoue, Hiromu Kishi, Satoru Kishida (Tottori Univ) NC2013-116
We investigated the effect of tiredness produced by a Kraepelin test on voice signals with the neural network system for... [more] NC2013-116
pp.163-166
EMM 2014-03-07
15:00
Ishikawa JAIST [Poster Presentation] Study on scramble method for speech signal by using random bit shift of quantization
Zhi Zhu, Katsuhiko Yamamoto, Masashi Unoki (JAIST), Naofumi Aoki (Hokkaido Univ.) EMM2013-109
Speech scrambling aims to eliminate the intelligibility of original speech in order to preventing eavesdropping and copy... [more] EMM2013-109
pp.57-62
SIS 2013-12-12
13:00
Tottori Torigin Bunka Kaikan (Tottori) [Tutorial Lecture] Enhancement and Separation for Speech Signals
Arata Kawamura (Osaka Univ.) SIS2013-35
In this paper, we discus about three main topics of speech processing technologies. First, we review and discuss about a... [more] SIS2013-35
pp.47-52
SIS 2013-12-12
13:50
Tottori Torigin Bunka Kaikan (Tottori) Correlation between voice signals and alcohol concentrations in blood
Masayuki Kawanoi, Naoki Fujiwara, Hiroki Yoshimura, Satoru Kishida (Tottori Univ.) SIS2013-37
We clarified the correlation between voice signals and alcohol concentrations in blood by using neural network systems f... [more] SIS2013-37
pp.59-62
SP 2013-01-30
15:45
Kyoto Doshisha Univ. A Study on Speaker Recognition Based on Decomposition of Periodic and Aperiodic Components
Yuki Ishikawa, Masafumi Nishida (Doshisha Univ.), Masakiyo Fujimoto (NTT), Seiichi Yamamoto (Doshisha Univ.) SP2012-102
In conventional researches, mel-frequency cepstral coefficients (MFCC) are widely used for a feature parameter which app... [more] SP2012-102
pp.25-30
EA, EMM 2012-11-17
15:00
Oita OITA Univ. A speech signal processing method by considering finite range of amplitude fluctuation and its application to actual environment
Akira Ikuta, Ran Xiao (Pref. Univ. Hiroshima), Kouji Hasegawa (Hiroshima Pref. Tec. Reseach Inst.), Mitsuo Ohta (Hiroshima Univ.) EA2012-107 EMM2012-89
The actual speech signal fluctuates within a finite range and the observed data sometimes display amplitude saturation o... [more] EA2012-107 EMM2012-89
pp.153-158
SP 2011-01-27
13:00
Kyoto NICT [Invited Talk] Recent advances of noise reduction technology in speech signal processing
Hiroshi Saruwatari (NAIST) SP2010-103
In this paper, I review basic principles and
recent advances of noise reduction methods used
in speech signal proces... [more]
SP2010-103
pp.1-6
EA 2010-11-19
11:00
Fukuoka Kyushu Univ. Noise Removal Method for Speech Recognition by Introducing Stochastic System Model
Akira Ikuta, Hisako Orimoto, Yuumi Nagami (Prefectural Univ. Hiroshima), Mitsuo Ohta (Hiroshima Univ, Emeritus) EA2010-95
Several noise removal methods for speech recognition have been proposed up to now. In this study, a method to remove the... [more] EA2010-95
pp.67-72
SP 2010-11-19
09:45
Aichi Aichi Prefectural Univ. Method of Speech Signal Decomposition into Periodic Component and Aperiodic Component Considering Changes in Pitch and Power
Masahiro Morita, Takehiko Kagoshima (Toshiba) SP2010-79
For the purpose of improving the voice quality of text-to-speech (TTS), we have been considering ways to decompose a spe... [more] SP2010-79
pp.59-64
RCS, SIP 2010-01-21
15:20
Fukuoka Kyushu Univ. A Lossless Data Hiding Method without Memorization of Hiding Parameter and Its Application to Image and Sound
Takahiro Tsuneyoshi, Masaaki Fujiyoshi, Hitoshi Kiya (Tokyo Metro. Univ.) SIP2009-86 RCS2009-220
This paper proposes a lossless data hiding method that is a parameter free method. For the hidden data extraction in the... [more] SIP2009-86 RCS2009-220
pp.77-82
SIS 2007-12-11
13:50
Hyogo   Impact Noise Suppression for Speech Signals by Using a Morphological Component Analysis with DFT
Hiroaki Hayashi, Makoto Nakashizuka, Youji Iiguni (Osaka Univ.) SIS2007-66
Morphological component analysis (MCA) is a signal separation method using sparse signal representations. For separation... [more] SIS2007-66
pp.47-52
IE, SIP 2006-04-21
15:30
Tokyo Kikai-Shinko-Kaikan Bldg. A Feedback Approach and Its Learning Algorithm for Overcomplete Blind Source Separation
Haruo Katou, Kenji Nakayama, Akihiro Hirano (Kanazawa Univ.)
A feedback structure and its learning algorithm are proposed for overcomplete blind source separation, where the number ... [more] SIP2006-9 IE2006-9
pp.49-54
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