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#1 |
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diyAudio Member
Join Date: Apr 2003
Location: Melbourne, Australia
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I am at uni doing electronics and one of our subjects is circuit and signal analysis. We are currently doing the sampling theorem and I am not sure on how this works.
How does one sample a signal then reconstruct it? We are using MATLAB to do this, so we just generate a sampled signal instead of sampling a continuous signal. From there, where do I go? All I can make out is that I Fourier transform it using FFT in MATLAB, then sample this function, run it through a low pass (ideal rectangle) filter then from there, I can reconstruct it, but I haven't the slightest clue how and I can't gather anythign from my text. http://www.jhu.edu/~signals/sampling/ I see this website does it. Any help would be great!! Thanks -Mikey |
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#2 | |
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diyAudio Member
Join Date: Dec 2001
Location: Germany
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Hi Mikey,
am in the same boat here. Just started to have a seminar called "signals&systems" working with Matlab also. Donīt know much about it but will try to help as far as I can. Quote:
The formulas of each step? Can you send me the m-file or did you simulate with Simulink? greets
__________________
jens |
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