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
Subject:
Noise Reduction and Speech Intelligibility
Click to purchase paper as a non-member or you can login as an AES member to see more options.
No AES members have commented on this paper yet.
To be notified of new comments on this paper you can subscribe to this RSS feed. Forum users should login to see additional options.
If you are not yet an AES member and have something important to say about this paper then we urge you to join the AES today and make your voice heard. You can join online today by clicking here.