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2015-03-20 21:17:18 +0200 | answered a question | Low pass filter for coherent demodulation? I found my mistake. In the lines I should not have passed an array of ones to This gives the correct output. On a side note, since the output of a linear system is effectively the convolution of its input and its unit impulse response, I could have used I recalled this after struggling with |

2015-03-19 16:12:23 +0200 | asked a question | Low pass filter for coherent demodulation? Hello, I'm trying to design a low-pass filter to recover a signal that was modulated using DSB-SC. I have produced the message signal and the modulated signal, but I can't seem to recover the original. To modulate, I use a 2 kHz sinusoid: dsb = m(t) * cos(2 Finally, I (try to) use an FIR low-pass filter with cutoff frequency 100 Hz. Here is what I'm doing: Based on the examples I've seen to implement filters, this looks to be the way it is done? Am I approaching this incorrectly? Ideally, the plots should look quite similar, but these don't. |

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2014-11-24 02:34:40 +0200 | commented answer | Condition in sum() function? This is precisely what I was looking for. I didn't even think to use |

2014-11-24 02:29:57 +0200 | commented question | Condition in sum() function? Yes, I only want to avoid n = 0. Thank you for your input, I ended up using the answer below. |

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2014-11-22 19:50:08 +0200 | asked a question | Condition in sum() function? Hello, I'm trying to use the exponential form of the Fourier series representation of a function to plot an approximation of said function using the first five terms. The actual function is f(t) = 1/2 + j/(2 Here I'm using 'E' to indicate summation notation. I apologize if this deviates from an established standard, but I'm having trouble uploading images right now (which would have made the function clearer). The code I'm using for this function is However, when I try this, I get the following exception:
This seems to be due to the division by |

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