dc.contributor.author | Hain, Thomas | en |
dc.contributor.author | Burget, Lukas | en |
dc.contributor.author | Dines, John | en |
dc.contributor.author | Garau, Giulia | en |
dc.contributor.author | Wan, Vincent | en |
dc.contributor.author | Karafiat, Martin | en |
dc.contributor.author | Vepa, Jithendra | en |
dc.contributor.author | Lincoln, Michael | en |
dc.date.accessioned | 2007-09-19T12:42:18Z | |
dc.date.available | 2007-09-19T12:42:18Z | |
dc.date.issued | 2007 | |
dc.identifier.citation | T. Hain, L. Burget, J. Dines, G. Garau, M. Karafiat, M. Lincoln, J. Vepa, and V. Wan. The AMI System for the Transcription of Speech in Meetings. In Proc. ICASSP, 2007 | |
dc.identifier.uri | http://hdl.handle.net/1842/2005 | |
dc.description.abstract | This paper describes the AMI transcription system for speech in
meetings developed in collaboration by five research groups. The
system includes generic techniques such as discriminative and speaker
adaptive training, vocal tract length normalisation, heteroscedastic
linear discriminant analysis, maximum likelihood linear regression,
and phone posterior based features, as well as techniques specifically
designed for meeting data. These include segmentation and
cross-talk suppression, beam-forming, domain adaptation, web-data
collection, and channel adaptive training. The system was improved
by more than 20% relative in word error rate compared to our previous
system and was used in the NIST RT’06 evaluations where it was
found to yield competitive performance. | en |
dc.format.extent | 116258 bytes | en |
dc.format.mimetype | application/pdf | en |
dc.language.iso | en | |
dc.subject | speech technology | en |
dc.subject | speech recognition | en |
dc.title | The AMI System for the Transcription of Speech in Meetings | en |
dc.type | Conference Paper | en |