Hi,
I am going to speculate that more fundamental issue is the ratio between bandwidth and the sampling rate.
The Nyquist Sampling theorem tell us the bandwidth is half the sampling frequency. Then you get folding and aliasing.
But that is an absolute minimum requirement.
As a
thought experiment consider this model:
It is a sample and hold circuit driven by a 1MHz sampling clock.
There is no need to worry about anti-aliasing filters because we are going to sample a pure sinewave so there is nothing to filter out.
The output of the sample and hold represents the data points that are stored in the scopes acquisition memory.
I have included a very simple reconstruction filter, a double pole filter at Fs/2.
This model helps us visualize the process.
I ran the model with different input frequencies.
100kHzThis is a fairly easy one. There are 10 samples of the input in the period and the sampling frequency is a multiple of the input frequency.
The circuit can construct a reasonable representation of the input signal from the samples.
So a ratio of 10 sample per period gives very good results.
107kHzThere are 9.34 samples per period, not a multiple.
You can still get a very reasonable representation of the input signal.
203KHzHere just less than 5 samples period. Injecting a pure sinewave, so anti-aliasing will not help.
There are barely enough data points to reconstruct the input signal.
303kHzStill above the Nyquist limit about 3.3 samples period.
There is some modulation caused by the sampling frequency.
If you look at V(sample) waveform, You have to guess at the input waveform from the sampled data.
There isn't enough data to create the waveform with any degree of confidence.
A 100Mhz scope with 500Msps isn't enough data points.
Regards,
Jay_Diddy_B