Literature DB >> 14759029

Performance of time- and frequency-domain binaural beamformers based on recorded signals from real rooms.

Michael E Lockwood1, Douglas L Jones, Robert C Bilger, Charissa R Lansing, William D O'Brien, Bruce C Wheeler, Albert S Feng.   

Abstract

Extraction of a target sound source amidst multiple interfering sound sources is difficult when there are fewer sensors than sources, as is the case for human listeners in the classic cocktail-party situation. This study compares the signal extraction performance of five algorithms using recordings of speech sources made with three different two-microphone arrays in three rooms of varying reverberation time. Test signals, consisting of two to five speech sources, were constructed for each room and array. The signals were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio of the output. A frequency-domain minimum-variance distortionless-response beamformer outperformed the time-domain based Frost beamformer and generalized sidelobe canceler for all tests with two or more interfering sound sources, and performed comparably or better than the time-domain algorithms for tests with one interfering sound source. The frequency-domain minimum-variance algorithm offered performance comparable to that of the Peissig-Kollmeier binaural frequency-domain algorithm, but with much less distortion of the target signal. Comparisons were also made to a simple beamformer. In addition, computer simulations illustrate that, when processing speech signals, the chosen implementation of the frequency-domain minimum-variance technique adapts more quickly and accurately than time-domain techniques.

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Year:  2004        PMID: 14759029     DOI: 10.1121/1.1624064

Source DB:  PubMed          Journal:  J Acoust Soc Am        ISSN: 0001-4966            Impact factor:   1.840


  6 in total

1.  Multi-microphone adaptive noise reduction strategies for coordinated stimulation in bilateral cochlear implant devices.

Authors:  Kostas Kokkinakis; Philipos C Loizou
Journal:  J Acoust Soc Am       Date:  2010-05       Impact factor: 1.840

2.  Localization of multiple acoustic sources with small arrays using a coherence test.

Authors:  Satish Mohan; Michael E Lockwood; Michael L Kramer; Douglas L Jones
Journal:  J Acoust Soc Am       Date:  2008-04       Impact factor: 1.840

3.  Adaptive spatial filtering improves speech reception in noise while preserving binaural cues.

Authors:  Susan R S Bissmeyer; Raymond L Goldsworthy
Journal:  J Acoust Soc Am       Date:  2017-09       Impact factor: 1.840

4.  Blind location and separation of callers in a natural chorus using a microphone array.

Authors:  Douglas L Jones; Rama Ratnam
Journal:  J Acoust Soc Am       Date:  2009-08       Impact factor: 1.840

5.  Two-microphone spatial filtering improves speech reception for cochlear-implant users in reverberant conditions with multiple noise sources.

Authors:  Raymond L Goldsworthy
Journal:  Trends Hear       Date:  2014-10-20       Impact factor: 3.293

6.  Real-time spectrum estimation-based dual-channel speech-enhancement algorithm for cochlear implant.

Authors:  Yousheng Chen; Qin Gong
Journal:  Biomed Eng Online       Date:  2012-09-24       Impact factor: 2.819

  6 in total

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