Dear Members of the Group,
since years I use a very simple method of measuring
the output sample rate of the soundcard:
(1) Choose an LF frequency normal such as HBG,
MSF or DCF77
(2) Generate a sinewave on the soundcard output
with a harmonic very near to the frequency
normal. Use maximum volme.
Example: For DCF77 on 77500 Hz generate
(77500 +- 1.5)/19 = 4078.868 or 4079.026 Hz
to get a signal 1.5 Hz apart from the normal
(if the samplerate has the nominal value).
(3) Use a diode or transistor to distort the sound
output and couple it appropriately to your LF
antenna such that the harmonic has similar
amplitude as the frequency normal in your
receiver.
(4) Change output frequency such that the frequency
normal lies symmetrically between both harmonics
(i.e. use both frequencies of the example in
sequence).
(5) Sample the SSB receiver output, compute the
spectrum, and identify the peaks. Usually the
generated peaks are far from what one expects.
So the audio frequencies have to be improved
iteratively.
The accuracy only is limited by the time interval
used for the measurements. With intervals of 10
minutes the accuracy is about 0.001 Hz. That is
enough to measure the drift of the samplerates.
I have some JPGs with measured spectra which will
be sent on request.
73 de Klaus, DJ5HG
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