I ask this question just because I'm curious to find out.
As far as I understand, the Nyquist theorem says that the sampling frequency must be at least twice the bandwidth of the signal to be sampled. So, if we want to sample a signal with a sample rate that is too low for its bandwidth, we have to low pass filter the signal prior to sampling to avoid aliasing. Or am I wrong?
My laptop has an integrated sound card. In Audacity I can select any arbitrary sample frequency. Does that mean, that my sound card contains an adjustable low pass filter at the input to filter the signal to the selected sampling rate?
How is such a variable filter designed in an integrated circuit?