DocumentCode
2701712
Title
The AMI System for the Transcription of Speech in Meetings
Author
Hain, Thomas ; Wan, Vincent ; Burget, Lukas ; Karafiat, Martin ; Dines, John ; Vepa, J. ; Garau, G. ; Lincoln, Mike
Author_Institution
Dept. of Comput. Sci., Sheffield Univ., UK
Volume
4
fYear
2007
fDate
15-20 April 2007
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 RT106 evaluations where it was found to yield competitive performance.
Keywords
maximum likelihood estimation; regression analysis; speech processing; AMI system; Web-data collection; beam-forming; channel adaptive training; cross-talk suppression; heteroscedastic linear discriminant analysis; maximum likelihood linear regression; meetings; phone posterior based features; speaker adaptive training; speech transcription; vocal tract length normalisation; word error rate; Ambient intelligence; Collaboration; Computer science; Error analysis; Hidden Markov models; Microphone arrays; NIST; Speech analysis; Speech recognition; Vocabulary; Meetings; Speech recognition;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech and Signal Processing, 2007. ICASSP 2007. IEEE International Conference on
Conference_Location
Honolulu, HI
ISSN
1520-6149
Print_ISBN
1-4244-0727-3
Type
conf
DOI
10.1109/ICASSP.2007.366923
Filename
4218111
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