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Time and Pitch Scale Modification of Audio Signals using Short Time Fourier Transform

Published on March 2013 by Srinivas Rao Chintagunta, Mrutyunjaya Nanda, Rajib Lochan Swain
International Conference on Computing, Communication and Sensor Network
Foundation of Computer Science USA
CCSN2012 - Number 3
March 2013
Authors: Srinivas Rao Chintagunta, Mrutyunjaya Nanda, Rajib Lochan Swain
9e3b6ea9-36e8-4820-9b32-a496fc0b8be6

Srinivas Rao Chintagunta, Mrutyunjaya Nanda, Rajib Lochan Swain . Time and Pitch Scale Modification of Audio Signals using Short Time Fourier Transform. International Conference on Computing, Communication and Sensor Network. CCSN2012, 3 (March 2013), 13-16.

@article{
author = { Srinivas Rao Chintagunta, Mrutyunjaya Nanda, Rajib Lochan Swain },
title = { Time and Pitch Scale Modification of Audio Signals using Short Time Fourier Transform },
journal = { International Conference on Computing, Communication and Sensor Network },
issue_date = { March 2013 },
volume = { CCSN2012 },
number = { 3 },
month = { March },
year = { 2013 },
issn = 0975-8887,
pages = { 13-16 },
numpages = 4,
url = { /specialissues/ccsn2012/number3/10862-1025/ },
publisher = {Foundation of Computer Science (FCS), NY, USA},
address = {New York, USA}
}
%0 Special Issue Article
%1 International Conference on Computing, Communication and Sensor Network
%A Srinivas Rao Chintagunta
%A Mrutyunjaya Nanda
%A Rajib Lochan Swain
%T Time and Pitch Scale Modification of Audio Signals using Short Time Fourier Transform
%J International Conference on Computing, Communication and Sensor Network
%@ 0975-8887
%V CCSN2012
%N 3
%P 13-16
%D 2013
%I International Journal of Computer Applications
Abstract

A method for independently modifying the time and pitch scale of acoustic signals, with an emphasis on speech signals is proposed in this paper. The algorithm developed here is based on Short Time Fourier Transform(STFT). The purpose of this paper is to devise a way to change the rate of a pre-recorded sound without altering the frequency content. Simply playing the sound at a different rate is not a solution. The frequencies would be distorted in proportion to the scaling factor, and at very low or high rates, would be very difficult to understand at all, let alone identify as human speech. Our approach is to sample the digital signal and then interpolate data points between our samples to produce a sound of the desired length. A slowed-down sound would have more points inserted between the samples than the original signal had, and a speeded-up sound would have fewer than the original. Performance of the proposed algorithm is demonstrated using spectrum plots.

References
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Index Terms

Computer Science
Information Sciences

Keywords

Speech Analysis Speech Processing Time Scale Modification Wavelet Packet Transform