An improved adaptive gain equalizer for noise reduction with low speech distortion
Blekinge Institute of Technology, School of Engineering, Department of Electrical Engineering2011 (English)In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 7Article in journal (Refereed) Published
In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.
Place, publisher, year, edition, pages
Springer , 2011. Vol. 7
Speech enhancement, Noise reduction, Noise-level estimation
IdentifiersURN: urn:nbn:se:bth-6967DOI: 10.1186/1687-4722-2011-7ISI: 000299122500005Local ID: oai:bth.se:forskinfoEF3499AC58836F03C125798A007408ECOAI: oai:DiVA.org:bth-6967DiVA: diva2:834529
Open Access article Article 72013-05-312012-01-192015-06-30Bibliographically approved