Open Access Research Article

Wavelet-Based Speech Enhancement Using Time-Frequency Adaptation

Kun-Ching Wang

Author Affiliations

Department of Information Technology & Communication, Shin Chien University, No. 200, University Road, Neimen Shiang, Kaohsiung 845, Taiwan

EURASIP Journal on Advances in Signal Processing 2009, 2009:924135  doi:10.1155/2009/924135


The electronic version of this article is the complete one and can be found online at: http://asp.eurasipjournals.com/content/2009/1/924135


Received: 22 February 2009
Revisions received: 21 July 2009
Accepted: 11 October 2009
Published: 1 December 2003

© 2009 The Author(s).

This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.

Abstract

Wavelet denoising is commonly used for speech enhancement because of the simplicity of its implementation. However, the conventional methods generate the presence of musical residual noise while thresholding the background noise. The unvoiced components of speech are often eliminated from this method. In this paper, a novel algorithm of wavelet coefficient threshold (WCT) based on time-frequency adaptation is proposed. In addition, an unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. The wavelet coefficient threshold (WCT) of each subband is first temporally adjusted according to the value of a posterior signal-to-noise ratio (SNR). To prevent the degradation of unvoiced sounds during noise, the algorithm utilizes a simple speech/noise detector (SND) and further divides speech signal into unvoiced and voiced sounds. Then, we apply appropriate wavelet thresholding according to voiced/unvoiced (V/U) decision. Based on the masking properties of human auditory system, a perceptual gain factor is adopted into wavelet thresholding for suppressing musical residual noise. Simulation results show that the proposed method is capable of reducing noise with little speech degradation and the overall performance is superior to several competitive methods.

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