MtPiNoY said:
... am not sure when to apply which to which situations. I am anxious to read your follow up post.
oh dear.
i do digital signal processing in audio and music for a living, so this is how
I think about it.
nearly always, when I'm thinking about a continuous-time audio signal and it's spectrum, i think in terms of the continuous Fourier Transform that is
X(f) = \int_{-\infty}^{\infty} x(t)\ e^{-j 2\pi f t}\ dt
i like that one because it is unitary
x(t) = \int_{-\infty}^{\infty} X(f)\ e^{+j 2\pi f t}\ df
so if i need to think in terms of the
Duality Property of the continuous Fourier Transform, it's trivial. no scaling needed. also convolution in either direction has no scalers. (however differentiation and integration and delay operations have an additional j2 \pi factor, but i think it's easier to remember that than to have to remember when to use 1/(2 \pi) in convolution.) in reality, signals that we have are in finite duration, so the F.T. can pretty well represent any practical signal.
i used to, when solving a transfer function for an analog filter, routinely substitute s \rightarrow j \omega as a shorthand. didn't mean that i was necessarily expressing it in terms of the Laplace Transform, but it looked like it. sometimes this was handy in doing partial fraction expansion.
but, if doing it in terms of s instead of j \omega, then you can talk about (complex valued) poles and zeros of a transfer function. that is normally in the context of the L.T.
What the Laplace Transform is good for, is the time-domain response to some transitioning input (like the step response). Sometimes time-domain behavior is more of interest to you than frequency domain. usually, for audio filters, i think only about the frequency response, but for a system like
"portamento" in a mono-phonic music synthesizer, then i am more concerned about the time-domain response. then it's L.T. because you can set up a problem with initial conditions, just as you would set it up to solve with differential equations. but it might be easier to do it with the L.T.
now, pretty much since the early 90's, any actual signal processing that i been writing code for, had to have been Digital Signal Processing, which is nearly always
discrete-time signal processing. now here, the thing to remember about is that periodicity in one domain implies discreteness of the other. unformly sampled signals have a spectrum that repeats every multiple of the sampling frequency. so we only need to think about the spectrum from -Nyquist to +Nyquist, which if we normalize goes from -\pi to +\pi which is the principal range in the Discrete-Time Fourier Transform (DTFT). that is really just the Fourier Transform applied to a discrete sequence and it has only terms that repeat every 2\pi. so you know that it's built-in to sampled data (which is what all we deal with in DSP), it's in its fundamental nature to mirror and repeat at \pm \pi. now since it is a discrete sequence, there are no differentiation or integral theorems, but there
is[/i] delay expressed as an operation and convolution is there, too.
The Z-transform's relationship to the DTFT is precisely the relationship of the Laplace Transform is to the continuous-time Fourier Transform. So if you were interested in constructing a time-domain response to a transient (like a step response), then doing it with the Z-transform is what you do. otherwise, for me it's just like another shorthand. but instead of
s \rightarrow j \omega
in the case of connecting the Laplace to Fourier regarding continuous-time signals...
... it's
z \rightarrow e^{j \omega}
which is what connects Z to DTFT regarding discrete-time signals.
so it really depends on whether you in a continuous-time or discrete-time environment, and what you're trying to do: deal with frequency-domain behavior or deal with time-domain behavior.