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

  • Authors:
  • Matthias Pawig;Gerald Enzner;Peter Vary

  • Affiliations:
  • Institute of Communication Systems and Data Processing, RWTH Aachen University, Aachen, Germany;Institute of Communication Acoustics, Ruhr-University, Bochum, Bochum, Germany;Institute of Communication Systems and Data Processing, RWTH Aachen University, Aachen, Germany

  • Venue:
  • IEEE Transactions on Signal Processing
  • Year:
  • 2010

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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.