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AUDIO SIGNAL PROCESSING

Platform : DSP

IEEE Projects Years : 2003

This paper appears in: Circuits and Systems, 2003. ISCAS '03. Proceedings of the 2003 International Symposium on Issue Date: 25-28 May 2003 AIM: TO UNDERSTAND SPEECH SIGNAL ENHANCEMENT TECHNIQE BASED ON MODIFICATION OF SPEECH FEATURES. ABSTRACT In audio processing, such as data compression and time/pitch scale modification, it is necessary to model the audio signal using an appropriate 'analysis' model. The 'sines+transient+noise' model, where the signal is separated into components of harmonic, transient and residual noise, is such a model that can be used in the analysis. In this model the harmonic separation is achieved by sending the signal through a uniform set of filters, usually having a fixed frequency resolution (bandwidth). In this paper we propose a pitch-synchronous IIR Laguerre filter having a time varying bandwidth (constant Q-factor) for the filter set. The advantages of using such a filter set in audio processing are discussed in the paper. LEARNING OBJECTIVE: To send the signal through a uniform set of filters, and to recover it after filtering INPUT: Original Speech signals and added it with noise signal OUTPUT : Recovered version of original speech signal APPLICATIONS: Speech enhancement in communication engineering SOFTWARE TOOL USED: MATLAB 2007A

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