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Brief Review of Fourier Analysis: Elena Punskaya
Brief Review of Fourier Analysis: Elena Punskaya
Elena Punskaya
www-sigproc.eng.cam.ac.uk/~op205
Some material adapted from courses by Prof. Simon Godsill, Dr. Arnaud Doucet, Dr. Malcolm Macleod and Prof. Peter Rayner
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Time domain Example: speech recognition difficult to differentiate between different sounds in time domain
tiny segment
sound /i/
as in see
sound /a/
as in father
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How do we hear?
Cochlea spiral of tissue with liquid and thousands of tiny hairs that gradually get smaller Each hair is connected to the nerve The longer hair resonate with lower frequencies, the shorter hair resonate with higher frequencies Thus the time-domain air pressure signal is transformed into frequency spectrum, which is then processed by the brain
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Inner Ear
www.uptodate.com
Fouriers Discovery
Jean Baptiste Fourier showed that any signal could be made up by adding together a series of pure tones (sine wave) of appropriate amplitude and phase
(Recall from 1A Maths)
Fourier Transform
The Fourier transform is an equation to calculate the frequency, amplitude and phase of each sine wave needed to make up any given signal :
Prism Analogy
Analogy:
a prism which splits white light into a spectrum of colors white light consists of all frequencies mixed together the prism breaks them apart so we can see the separate frequencies
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White light
Spectrum of colours
Signal
Fourier Transform
Spectrum
Signal Spectrum
Every signal has a frequency spectrum. ! the signal defines the spectrum ! the spectrum defines the signal We can move back and forth between the time domain and the frequency domain without losing information
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spectrum
!"
!"
sound /a/
as in father
!"
in logarithmis units of dB
peaks correspond to the resonances of the vocal tract shape they can be used to differentiate between sounds
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sound /i/
as in see
t
in logarithmis units of dB
!"
using the sifting property of the !-function to reach the last line
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The signal sample values may be expressed in terms of DTFT by noting that the equation above has the form of Fourier series (as a function of ") and hence the sampled signal can be obtained directly as
[You can show this for yourself by first noting that (*) is a complex Fourier series with coefficients however it is also covered in one of Part IB Examples Papers]
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The DTFT
expresses the spectrum of a sampled signal in terms of the signal samples but is not computable on a digital computer for two reasons: 1.! The frequency variable " is continuous. 2.! The summation involves an infinite number of samples.
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The inverse DFT can be used to obtain the sampled signal values from the DFT: multiply each side by and sum over p=0 to N-1
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! !
[You should check that you can show these results from first principles]
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!=2#f$ = 2#(f/fs) -
f - cycles per second fs - samples per second f/fs - cycles per sample
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