Abstract
This project presents an implementation of audio compression using the Fast Fourier Transform (FFT). The fundamental concept is to find innovative ways to reduce the data rate of the audio signal to as low as possible while still keeping the signal intelligible. In this project Fast Fourier Transform and Inverse Fast Fourier Transform are used for the compression and decompression of a speech signal. This audio compression scheme is simulated using MATLAB 6.5. Simulations are performed for different Fast Fourier Transform sizes and different number of components chosen. Two different methods were used that are by retaining the first n-components and by retaining dominant n-components. The Signal-To-Noise-Ratios are computed for all the simulations and used to study the behaviours of the compression scheme using Fast Fourier Transform. The noise introduced in the signal (for various cases) is studied both by listening to the recovered signal and by the calculated Signal-To-Noise-Ratios.
Metadata
Item Type: | Thesis (Degree) |
---|---|
Creators: | Creators Email / ID Num. Abdul Mutillip, Norbaizura UNSPECIFIED |
Contributors: | Contribution Name Email / ID Num. Thesis advisor Khadri, Norasimali UNSPECIFIED |
Subjects: | T Technology > TK Electrical engineering. Electronics. Nuclear engineering > Electronics > Apparatus and materials T Technology > TK Electrical engineering. Electronics. Nuclear engineering > Electronics > Applications of electronics |
Divisions: | Universiti Teknologi MARA, Shah Alam > Faculty of Electrical Engineering |
Programme: | Bachelor of Electrical Engineering (Hons) |
Keywords: | Audio compression, audio signal, Fourier Transform (FT) |
Date: | 2003 |
URI: | https://ir.uitm.edu.my/id/eprint/67863 |
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