bibtype C - Conference Paper (international conference)
ARLID 0348108
utime 20240111140745.2
mtime 20101102235959.9
title (primary) (eng) Adaptive Time-Domain Blind Separation of Speech Signals
specification
page_count 8 s.
media_type memory stick
serial
ARLID cav_un_epca*0348107
ISBN 978-3-642-15994-7
title Lecture Notes in Computer Science
part_num 6365
page_num 9-16
publisher
place Heidelberg
name Springer-Verlag
year 2010
editor
name1 Gavrilova
name2 M.L.
editor
name1 Kumar
name2 V.
editor
name1 Mun
name2 Y.
editor
name1 Tan
name2 C.J.K.
editor
name1 Gervasi
name2 O.
keyword blind source separation
keyword speech
keyword convolutive mixture
keyword adaptive algorithms
author (primary)
ARLID cav_un_auth*0050739
name1 Málek
name2 J.
country CZ
author
ARLID cav_un_auth*0108100
name1 Koldovský
name2 Zbyněk
full_dept (cz) Stochastická informatika
full_dept Department of Stochastic Informatics
department (cz) SI
department SI
institution UTIA-B
full_dept Department of Stochastic Informatics
fullinstit Ústav teorie informace a automatizace AV ČR, v. v. i.
author
ARLID cav_un_auth*0101212
name1 Tichavský
name2 Petr
full_dept (cz) Stochastická informatika
full_dept Department of Stochastic Informatics
department (cz) SI
department SI
institution UTIA-B
full_dept Department of Stochastic Informatics
fullinstit Ústav teorie informace a automatizace AV ČR, v. v. i.
source
url http://library.utia.cas.cz/separaty/2010/SI/tichavsky-adaptive time-domain blind separation of speech signals.pdf
source_size 281kB
cas_special
project
project_id 1M0572
agency GA MŠk
ARLID cav_un_auth*0001814
project
project_id GA102/09/1278
agency GA ČR
ARLID cav_un_auth*0253174
project
project_id GA102/08/0707
agency GA ČR
country CZ
research CEZ:AV0Z10750506
abstract (eng) We present an adaptive algorithm for blind audio source separation (BASS) of moving sources via Independent Component Analysis (ICA) in time-domain. The method is shown to achieve good separation quality even with a short demixing filter length (L = 30). Our experiments show that the proposed adaptive algorithm can outperform the off-line version of the method (in terms of the average output SIR), even in the case in which the sources do not move, because it is capable of better adaptation to the nonstationarity of the speech.
action
ARLID cav_un_auth*0264720
name Latent Variable Analysis and Signal Separation
place St. Malo
dates 27.09.2010-30.09.2010
country FR
reportyear 2011
RIV BB
permalink http://hdl.handle.net/11104/0188720
arlyear 2010
mrcbU56 281kB
mrcbU63 cav_un_epca*0348107 Lecture Notes in Computer Science 6365 978-3-642-15994-7 9 16 Heidelberg Springer-Verlag 2010 Latent Variable Analysis and Signal Separation, 9th Int. Conf., LVA/ICA 2010
mrcbU67 Gavrilova M.L. 340
mrcbU67 Kumar V. 340
mrcbU67 Mun Y. 340
mrcbU67 Tan C.J.K. 340
mrcbU67 Gervasi O. 340