DSP very basic question

[shhrikant1]

Correct me if I am wrong -
If we have a continuous signal & we have to sample it ,
then -
impulse train of sampling frequency F(s) * Continuous signal X(t)
will give me sampled output X(s).

here * is convolution.

also by going through convolution theory, I am quite clear that convolution is totally different from multiplication in time domain, then how to write a code for convolution in VHDL?

I am very new to DSP , as you could guess by the simplicity of the question itself.

thanks in advance

 

[permute]

In this case, you are talking about a mathmatical model of sampling. so you will just be sampling some input signal.

after sampling, you might choose to use a FIR filter or an IIR filter. both perform a convolution defined by the filter.

 

[shhrikant1]

Hi Permute,

I am sorry I couldn't understand this line " after sampling, you might choose to use a FIR filter or an IIR filter. both perform a convolution defined by the filter
"
I thought I have to do convolution for sampling, What I could infer from the above statement is sampling is done 1st & by designing a filter convolution is done by the filter itself. plz correct me, its something very basic, which I couldn't answer myself

 

[Ahmed Alaa]

Sampling is the multiplication of the continuous signal by an impulse train in the TIME domain, the convolution is applied between the signal's spectrum and an impulse train to obtain the output's spectrum.

If you are to do sampling, there is no need for convolution at all !