Volume 10, Issue 1 (9-2013)                   JSDP 2013, 10(1): 68-57 | Back to browse issues page

XML Persian Abstract Print

Download citation:
BibTeX | RIS | EndNote | Medlars | ProCite | Reference Manager | RefWorks
Send citation to:

Kittler J. Speech Signals Processing Using New Compressive Sampling Based Feature Extraction Method . JSDP. 2013; 10 (1) :68-57
URL: http://jsdp.rcisp.ac.ir/article-1-63-en.html
Abstract:   (12178 Views)
In this paper, we present a Compressive Sampling (CS)-based feature extraction method for audio signals. In the proposed approach, the audio signal is firstly segmented by hamming windows and the Discrete Fourier Transform (DFT) of the samples is calculated within each frame. Then, the normalized values of the DFT coefficients of each frame are accumulated. At the next step, the second DFT is applied on the vector formed from the accumulated sum in consecutive frames. Finally, considering the sparseness of the resulted vector, our proposed CS-2FFT feature vector is achieved by a random sampling. In this research, the performance of CS-2FFT feature vector has been examined in the applications of audio classification and audio source localization. The simulation show that the proposed feature vector results in a classifier which is more accurate and less computationally complex compared to the classical classifiers. Also, it is shown that the employing CS-2FFT feature vector, the localization error will be less than 2%.
Full-Text [DOCX 66 kb]   (10291 Downloads)    
Type of Study: Research | Subject: Paper
Received: 2013/06/3 | Accepted: 2013/10/12 | Published: 2013/12/3 | ePublished: 2013/12/3

Add your comments about this article : Your username or Email:

© 2015 All Rights Reserved | Signal and Data Processing