I always studied the DFT starting from his formula, but for some reasons I need to do comparison between the FT and the DFT. I found the pdf in this link very useful http://www.robots.ox.ac.uk/~sjrob/Teaching/SP/l7.pdf because it explains how the DFT can be obtained starting from the FT. After the first formula it says " We could regard each sample f[k] as an impulse having area f[k]". What I don't understand is how it possible that the sample, being an impulse, has an area f[k]. Shouldn't t have an area f[k] multiplied by a very low number? This passage for me is crucial because I cannot understand why, passing from the FT to the DFT, the time increment disappears.
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The area under an impulse function is one. So if you multiply a continuous time waveform with an impulse, the result will be a weighted impulse with the area equal to whatever the waveform was at that moment in time.
Dan Boschen
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T = 1/Fs; % Sampling period
L = 1500; % Length of signal t = (0:L-1)*T; % Time vector
S = 0.7sin(2pi50t) + sin(2pi120*t);
f = Fs*(0:(L/2))/L;
Y = fft(S); P2 = abs(Y/L); P1 = P2(1:L/2+1); P1(2:end-1) = 2*P1(2:end-1);
plot(f,P1) title('Single-Sided Amplitude Spectrum of S(t)') xlabel('f (Hz)') ylabel('|P1(f)|')
– Ashish Bhigah Jan 05 '20 at 20:27code– jomegaA Feb 06 '20 at 08:45Is this a general assumption or specific to this problem here?
So the scaling is necessary when approximating CT Fourier transfrom by the DFT? In other post you have nice proof for it.
– jomegaA Feb 06 '20 at 09:19