DocumentCode
1476448
Title
Adaptive co-channel speech separation and recognition
Author
Yen, Kuan-Chieh ; Zhao, Yunxin
Author_Institution
Dept. of Electr. & Comput. Eng., Illinois Univ., Urbana, IL, USA
Volume
7
Issue
2
fYear
1999
fDate
3/1/1999 12:00:00 AM
Firstpage
138
Lastpage
151
Abstract
An improved technique of co-channel speech separation, S-AADP/LMS, and its integration with automatic speech recognition is presented. The S-AADF/LMS technique is based on the algorithms of accelerated adaptive decorrelation filtering (AADP) and LMS noise cancellation, where a switching between the two algorithms is made depending upon the active/inactive status of the co-channel signal sources. The AADF improves the previous adaptive decorrelation algorithm in terms of system stability and estimation efficiency, and leads to better estimation of time-varying and reverberant channels. The S-AADF/LMS further improves the estimation accuracy when only one source signal remains active during certain periods of time. A coherence-function based source signal detection algorithm is also presented, which is successfully used in the switching between AADF and LMS and in extracting speech signals from leakage-corrupted background. Experiments were conducted under a simulated environment based on the measurements made of certain real room-acoustic conditions, and the results demonstrated the effectiveness of the proposed technique for co-channel speech separation and recognition
Keywords
adaptive filters; adaptive signal detection; adaptive signal processing; architectural acoustics; decorrelation; filtering theory; least mean squares methods; noise abatement; speech processing; speech recognition; S-AADP/LMS; accelerated adaptive decorrelation filtering; active/inactive status; adaptive co-channel speech separation; adaptive decorrelation algorithm; automatic speech recognition; co-channel signal sources; coherence-function; estimation accuracy; estimation efficiency; experiments; leakage-corrupted background; noise cancellation; reverberant channels; room acoustics; simulated environment; source signal detection algorithm; system stability; time-varying channels; Acceleration; Adaptive filters; Automatic speech recognition; Decorrelation; Filtering algorithms; Least squares approximation; Noise cancellation; Speech recognition; Stability; Time varying systems;
fLanguage
English
Journal_Title
Speech and Audio Processing, IEEE Transactions on
Publisher
ieee
ISSN
1063-6676
Type
jour
DOI
10.1109/89.748119
Filename
748119
Link To Document