DocumentCode
285002
Title
Fast echo cancellation in a voice-processing system
Author
Raman, Vijay R. ; Cromack, Mark R.
Author_Institution
Digital Sound Corp., Carpinteria, CA, USA
Volume
4
fYear
1992
fDate
23-26 Mar 1992
Firstpage
513
Abstract
An efficient adaptive echo cancellation scheme where near-end hybrid echoes are considered the dominant factor in poor DTMF detection and speech recognition in the presence of outbound speech is described. The incorporation of distinct and separate adapt and cancel filters is crucial in attaining significant echo reduction. The adapt filter performs nonreal-time identification of the echo transfer function with buffered transmit and receive data that meet specific criteria. The adaptation implementation is controlled by what has been termed cycle steal; after the required functions have been performed, the remaining real time is utilized for adaptation. After each convergence interval, a windowed set of coefficients is passed from the adapt filter to the cancel filter. The cancel filter performs echo cancellation in real time with the defined coefficients and produces dramatic improvements in measured DTMF detection performance
Keywords
acoustic signal processing; adaptive filters; convergence; digital filters; echo suppression; identification; speech analysis and processing; speech recognition; adapt filter; adaptive echo cancellation; cancel filter; convergence interval; echo transfer function; near-end hybrid echoes; nonreal-time identification; outbound speech; poor DTMF detection; speech recognition; voice-processing system; Acoustic noise; Circuits; Convergence; Detectors; Echo cancellers; Filters; Performance evaluation; Speech recognition; Telephony; Transfer functions;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech, and Signal Processing, 1992. ICASSP-92., 1992 IEEE International Conference on
Conference_Location
San Francisco, CA
ISSN
1520-6149
Print_ISBN
0-7803-0532-9
Type
conf
DOI
10.1109/ICASSP.1992.226398
Filename
226398
Link To Document