Hi, I am new to filter design and I have designed an equiripple filter in matlab using the commands:
With a passband frequency of 30MHz and a stopband frequency of 50MHz. However, when I filter the signal
with the command
both sine-functions are reduced. But sin(100*x) should go through and I don't understand why it doesn't. Does anyone know what I am doing wrong?
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Hi Jenny, you don't tell me what Fs is here so I cannot say whether or not your design is correct.
Is Fs 10 kHz? in your time vector? if so then why are you stepping in increments of 1/(4*Fs)?? That changes the sampling rate. Your time increments have to be 1/Fs by definition.
Fs = 1e4; t = 0:1/Fs:1-1/Fs; y=sin(1000*2*pi*t)+cos(2*pi*250*t); % design filter Hd=fdesign.lowpass('Fp,Fst,Ap,Ast',500,700,0.5,60,10000); d=design(Hd,'equiripple'); out = filter(d,y); outdft = fft(out); plot(abs(outdft))
The sampling rate you use for the signal has to match the sampling rate used in the filter design.
I don't see that with your unit step time vector, your frequencies in
y = 2+sin(1000*x)+5*sin(1e9*x)
are appropriate. What frequency do you think sin(1000*x) is?
Your x vector has unit steps. If we translate this to your sampling frequency (used in your filter design) of 1 gigahertz, then these frequencies are aliased.
f = 1000/(2*pi)
Now with a unit step that means it has a period less than 1 sample. The period of sin(1000*x) is (2*pi)/1000. That's above the Nyquist considerably and therefore so is sin(10^9*x) even more so.
Why are you defining your signal like this? Why not use a time vector where you use the sampling interval that corresponds to your sampoing frequency?
Fs = 1e9; t = 0:1/Fs:(1e5*(1/Fs))-(1/Fs);
I just set up t to be the same length as your original x.
Now define your sine waves over the t vector.
Thank you very much Wayne!
Your reply helped me generate a signal with the frequency I intended it to have, I didn't think about aliasing. However, when I create a signal (with the correct frequency), that should be damped, it's still there. So I wonder if I am still doing something wrong with the signal-generating. I decided to work with lower frequency levels (just to make it a bit easier) and I have generated a filter and a signal with the following commands:
t = 0:1/(4*Fs):(100*(1/(Fs)))-(1/(Fs)); y=sin(1000*2*pi*t);
Then I filtered the signal using the command
But I still see a signal with amplitude 1, shouldn't it be reduced since the 1000 Hz-signal is above the frequency stop of 700?
Oh, yes, now I can see that it works, thank you!
I did not now that the sampling rates had to be the same, why is that? I decided to divide it by four because the generated signal became smoother when I worked with higher frequencies, but I guess I just have to raise the sampling rate in that case?