In real world we sample a signal from A/D suppose with Fs=100 Hz..Now when we see that
digital signal its non uniform sampled.
Now, on non uniform signal Please correct me if am wrong:)
i want to apply oversampling which means that i need to sample it again with Fs=400Hz....and later regularise it. But how will i apply oversampling on it.....I mean i dont have amplitude value or i can say that i dont have signal to sample i only have some non uniform discrete values
How over sampling is apply in the real world.
1. We always do over sampling during A/D converter
2. Also how can i apply window function, I didnt get reply for this
1.if my orginal signal s is between -30 to 30 Hz 40 by 1
2.And signal on whch filter need to apply is 127 by 1 w
3. Now how can i recover s by applying rectangular window function on w. I mean dimensions are different..its not working on matlab
I am doing
"guj " <gulatiakshay@gmail.com> wrote in message <h40iv9$2au$1@fred.mathworks.com>...
> I really need to clear this :),
>
> In real world we sample a signal from A/D suppose with Fs=100 Hz..Now when we see that
> digital signal its non uniform sampled.
>
> Now, on non uniform signal Please correct me if am wrong:)
>
> i want to apply oversampling which means that i need to sample it again with Fs=400Hz....and later regularise it. But how will i apply oversampling on it.....I mean i dont have amplitude value or i can say that i dont have signal to sample i only have some non uniform discrete values
>
> How over sampling is apply in the real world.
>
> 1. We always do over sampling during A/D converter
> 2. Also how can i apply window function, I didnt get reply for this
>
> 1.if my orginal signal s is between -30 to 30 Hz 40 by 1
> 2.And signal on whch filter need to apply is 127 by 1 w
> 3. Now how can i recover s by applying rectangular window function on w. I mean dimensions are different..its not working on matlab
> I am doing
>
> w.*RECTWIN(SIZE(s))
>
> Any help will be appreciated
> Thanks
>
> Aki
I think, I need to put my doubt in right way :)
dft can be defined in a way
P(m) = sum{n=1:N} p(n) exp(j * m * delk*x(n))
delk=2*pi/T
m=-M/2-----M/2-1
Now my question is i have N irregular spaced signals....and I am Transforming N samples to M points
oversampling=Nc/M;
now when oversampling is 2
Nc=2M, so i am applying FFT on Nc samples, But where are my Nc samples i only have
N samples....how can i apply FFT on Nc signal when i dont have N values...
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