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TSP
2010

Adaptive sampling rate correction for acoustic echo control in voice-over-IP

9 years 6 months ago
Adaptive sampling rate correction for acoustic echo control in voice-over-IP
Abstract--Hands-free terminals for speech communication employ adaptive filters to reduce echoes resulting from the acoustic coupling between loudspeaker and microphone. When using a personal computer with commercial audio hardware for teleconferencing, a sampling frequency offset between the loudspeaker output D/A converter and the microphone input A/D converter often occurs. In this case, state-of-the-art echo cancellation algorithms fail to track the correct room impulse response. In this paper, we present a novel least mean square (LMS-type) adaptive algorithm to estimate the frequency offset and resynchronize the signals using arbitrary sampling rate conversion. In conjunction with a normalized LMS-type adaptive filter for room impulse response tracking, the proposed system widely removes the deteriorating effects of a frequency offset up to several Hz and restores the functionality of echo cancellation.
Matthias Pawig, Gerald Enzner, Peter Vary
Added 22 May 2011
Updated 22 May 2011
Type Journal
Year 2010
Where TSP
Authors Matthias Pawig, Gerald Enzner, Peter Vary
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