(New page: For this question, the original signal is a square wave with maximum amplitude one, minimum amplitude zero and period P. The ideal low pas filter is a band that goes from -fc to fc with c...) |
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From here, I am not sure what to do. I feel as if the amplitude should be adjusted by 1/period, but I am not sure. Currently, we are in the time domain, but the digital filter is in the frequency domain. How do you apply this filter? Any ideas are helpful. Thanks! | From here, I am not sure what to do. I feel as if the amplitude should be adjusted by 1/period, but I am not sure. Currently, we are in the time domain, but the digital filter is in the frequency domain. How do you apply this filter? Any ideas are helpful. Thanks! | ||
--[[User:Babaumga|Babaumga]] 12:27, 11 February 2009 (UTC) | --[[User:Babaumga|Babaumga]] 12:27, 11 February 2009 (UTC) | ||
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| + | I know this is late but this really helped visualize what your sampler is doing | ||
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| + | [[Getting X(w) from X(f)]] | ||
Latest revision as of 17:26, 11 February 2009
For this question, the original signal is a square wave with maximum amplitude one, minimum amplitude zero and period P. The ideal low pas filter is a band that goes from -fc to fc with constant amplitude one. Once filtered (for part a), the signal is once period of the original signal that starts at -1e-4 and goes to 1e-4. The ideal sampler creates a discrete signal with 5 points each 5e-5 apart.
From here, I am not sure what to do. I feel as if the amplitude should be adjusted by 1/period, but I am not sure. Currently, we are in the time domain, but the digital filter is in the frequency domain. How do you apply this filter? Any ideas are helpful. Thanks! --Babaumga 12:27, 11 February 2009 (UTC)
I know this is late but this really helped visualize what your sampler is doing
