We present a new algorithm for real-time noise reduction of audio signals. In order to derive the noise reduction function, the proposed method adaptively estimates the instantaneous noise spectrum from an autoregressive signal model as opposed to the widely-used approach of using a constant noise spectrum fingerprint. In conjunction with the Ephraim and Malah suppression rule a significant reduction of both stationary and non-stationary noise can be obtained. The adaptive algorithm is able to work without user interaction and is capable of real-time processing. Furthermore, quality improvements are easily possible by integration of additional processing blocks such as transient preservation.

Authors: Wiesener, Constantin; Flohrer, Tim; Lerch, Alexander; Weinzierl, Stefan
Affiliations: TU Berlin, Berlin, Germany; zplane.development, Berlin, Germany(See document for exact affiliation information.)
AES Convention: 128 (May 2010) Paper Number: 8048
Publication Date: May 1, 2010 Import into BibTeX
Subject: Noise Reduction and Speech Intelligibility
Permalink: https://www.aes.org/e-lib/browse.cfm?elib=15345

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