راهکار جدید استخراج ویژگی مبتنی بر نمونه‌برداری فشرده در پردازش سیگنال‌های صوتی

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%.