On Sun, 5 Aug 2001 22:46:14 +0300, "Johan Bodin"
<[email protected]> wrote:
I found an interesting tutorial article about undersampling and ADC
Since the article is from 1995, the list of available components is
not up to date. On the EDN site, there are also other related
articles, if you enter "undersampling" into the search functionality.
Fs=6kHz seems like a good choice. The selected Fs harmonic should be just
the band of interest (lowest acceptable beat note) and the preselector BW
must be less than Fs/2. Pse correct me if I'm wrong.
This is true as this Fs/2 bandwidth is exactly at nFs .. (n+0.5)Fs or
at (n+0.5)Fs .. (n+1)Fs. At any other frequency bands, the filter
should be narrower in order to avoid the nFs and (n+0.5)Fs
By the way, standard audio ADCs preceded by a sample and hold circuit
might also be usable. When running at 48 kHz Fs, the multiples would
be at 96 and 144 kHz, thus the 135.7 .. 137.8 kHz band would be
converted to 8.3 .. 6.2 kHz (inverted spectrum). With such high
sampling rate, the front end filtering could be done directly at
signal frequency using ordinary LC filters. The most problematic alias
is 150.2 .. 152.3 kHz, which would require a deep notch. The other
close alias is at 102.2 .. 104.3 kHz, but apparently this would be
sufficiently attenuated by the stop band of any bandpass filter.
The other advantage of using standard audio ADCs producing SPDIF is
that sound cards with SPDIF inputs could be used to input the digital
signal into PC for further processing.
The nasty thing about these converters is that they usually require a
clock frequency of 64Fs .. 256Fs, which would be quite problematic to
generate at very high precision, if 48 kHz sampling rate is used. The
converter might be clocked to run at 50 kHz, simplifying the reference
frequency generation, but the PC sound card might not lock on to the
50 kHz sample rate.