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[Linrad] Re: polarization measurement with digital downconversion receivers
- Subject: [Linrad] Re: polarization measurement with digital downconversion receivers
- From: Leif Asbrink <sm5bsz.com; leif@xxxxxxxxxxxxxxxx>
- Date: Sun, 22 Feb 2009 22:37:20 +0100
> Forgive me - I do not know how Linrad figures the polarization (is it described on your web site? or is it Joe Taylor's?). It seems to me that Linrad has two signals: one that is known to be vertical and one that is known to be horizontal. I expect that the only phase angle you could measure reliably would be one versus the other - i.e., one is leading or lagging the other. So you might determine that the received signal is (for example) 23 degrees clockwise from the vertical towards the horizontal.
> This 140 degree phase error (or shift) that you describe would apply to both signals - so you would still find that the received signal was "23 degrees clockwise from the vertical towards the horizontal" even with the 140 degree shift - no?
No. It would not, except at the frequency where you calibrated. A time shift
is the same as a phase shift, The correct phase shift at 144.100 MHz might be
302468 degrees. When you calibrate the angle you correct for is something
between 0 and 360 degrees. The missing information about perhaps a thousand
full turns is the fundamental problem. When you move just 50 kHz, the phase shift
from those has to be accounted for, otherwise you get an unacceptable
> BTW - I believe?that we can sync the CIC filters etc. in the Mercury boards, so I am hoping to have something that works in that way eventually. (Travel in Europe and Asia this month has me away from the shack for an extended amount of time). I am thinking that the first experiment I?would do would be to take the?96 kHz baseband from the audio outputs on the Mercury boards and feed them into a sound card like the Delta44. That way (I hope) no changes would be required on the Linrad MAP65 side of things.
> That would completely remove the issue of making the twin USB ports work (for now anyway).?
Yes. With a signal generator and an oscilloscope you can see the two
audio signals and make sure they have a constant phase relation.
You may then change the signal generator frequency and see how
the phase angle varies with frequency.
If you reset one FPGA you should see the signal come back in a different
phase that has a different behaviour with frequency on its phase
relation to the other channel.
Presumably the analog sound output comes directly from the FPGA and
that would be very fine because then you would have the FPGA sync
problem isolated for a first level of testing.
Leif / SM5BSZ
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