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The Nyquist / Shannon sampling theorem is like super famous in signal processing. It says that when doing regularly spaced point-wise sampling we need to sample a sinusoidal signal at least two times each period to avoid frequency aliasing.

But what happens if we don't just sample the function value in instants in time, but say a local Taylor approximation?

$$f_k(x) = \sum_{l=0}^Nc_{kl}(x-k\Delta_x)^l$$

For $N=0$ this becomes the usual sampling $f_k(x) = c_{k0}$, we only measure the function values but none of the derivatives.

Below is an illustration of Nyquist phenomenon on a typical chirp signal. We see the catastrophy that occurs when the local frequency increases above the sampling rate prescribed by Nyquist. But what would happen if we could measure also the slope at the green points or even the second derivatives et.c.? Could we push the bound upwards?

enter image description here

mathreadler
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  • How do you represent a Taylor approximation in the frequency domain? – AnonSubmitter85 Mar 19 '18 at 17:00
  • @AnonSubmitter85 good question, i suppose that is part of the problem. – mathreadler Mar 19 '18 at 18:02
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    If $x(t)$ is a time-domain signal and we consider only the first derivative, rather than having just $X(j\omega)$ convolved with a comb function (where the spacing of the impulses is the reciprocal of the sampling period), we also know $j\omega X(j\omega)$ convolved with the same comb function. Does that sound right? I am not sure. – AnonSubmitter85 Mar 20 '18 at 04:40
  • I am open to any approach, I was mostly curious to see how my numerical solver would meet the theoretical bounds which anyone might show / know about. – mathreadler Mar 20 '18 at 06:58
  • Yes it sounds like a reasonable approach. Would it be in general to know $(j\omega)^kX(j\omega)$ for k:th derivative? It was what I was guessing that the monomials correspond to the derivatives, (or the solutions to linear differential equations would not be roots to a polynomial, which they are because we did that a lot in uni). – mathreadler Mar 20 '18 at 07:06

1 Answers1

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I don't claim this is an answer. I am more thinking out loud here and this is too much for a comment.

The simplest proof of the sampling theorem is to show that any signal with frequency less than some cut-off frequency, $\vert f \vert < f_c $, can be reconstructed by applying an ideal low-pass filter (defined by the same cut-off frequency) to the sampled version of the sinusoid. We can represent the sampling at a frequency of $f_s=1/T_s$ in the time domain by multiplying the input signal $x(t)$ by

$$ \textrm{comb}_{T_s}(t) = \sum_{k}{\delta(t-kT_s) }. $$

In the frequency domain this becomes

$$ x(t) \cdot \textrm{comb}_{T_s}(t) \leftrightarrow X(f) \ast \textrm{comb}_{f_s}(f). $$

Since the replicas will be separated by $f_s$ Hertz, if $X(f)$ has zero magnitude for $\vert f\vert > f_s/2$ and we let $f_c = f_s/2$, then

$$ \textrm{rect}\left( f \over 2f_c \right) \cdot \left( X(f) \ast \textrm{comb}_{f_s}(f) \right) = X(f). $$

Thus, if $x(t)$ has no spectral content at frequencies less than $-f_s/2$ and greater than $+f_s/2$, then the sampling frequency $f_s$ is sufficient to reconstruct it.

If we instead sampled the first derivative of $x(t)$, then we would have

$$ \left( { {d} \over {dt} } x(t) \right) \cdot \textrm{comb}_{T_s}(t) \leftrightarrow \left( f \cdot X(f) \right) \ast \textrm{comb}_{f_s}(t). $$

Suppose that $\vert f_1 \vert < f_s/2$ and $\vert f_2 \vert < f_s$ and $\vert f_1 - f_2 \vert = f_s$. If $x(t) = a \cdot e^{j2\pi f_1 t} + b \cdot e^{j2\pi f_2 t}$, $a,b\in\mathbb{C}$, and we sample both the value and first derivative at a rate of $f_s$, then the spectrums we see after applying the low-pass filters are

$$ X_{f_s}(f) = a\cdot\delta(f-f_1) + b\cdot\delta(f-f_1) \\ fX_{f_s}(f) = a\cdot f_1 \cdot \delta(f-f_1) + b \cdot f_2 \cdot\delta(f-f_1) , $$

which, if I am not mistaken, is enough information to reconstruct $x(t)$ given the restrictions on $f_1$ and $f_2$. So having the sample value as well as the first derivative lets us double the frequency below which we can accurately reconstruct the time domain signal.

Far from a proof, I know. But I thought the question was interesting and wanted to think about it.