Re Question For Craig Markwardt

"George Dishman" wrote in message
...

"ralph sansbury" wrote in message
...
George, I know you are a superior EE and that

We are talking about the size of the intermediate frequency
range
relative
to the original range 1MHz is small relative to 200MHz but
not
to 1Hz

The terms "narrow band" and "wide band"

But the size of the intermediate frequency relative to the
original
range is what we are talking about. You seem to have your own
subjective
read of what others say and on what you say without understanding
that words are ambiguous and you have to say out loud what you
mean
or what you think is meant before going off halfcocked.


compare the width
of the equipment to the width of the signal being processed.
Wide band in this context means sufficiently wide that it
does not exclude any frequency of interest or produce any
modification of the signal such as emphasising one frequency
more than another.

sequence of voltages at this difference
frequency.(mixer
and
repeated heterodyne up and down conversion etc is the
jargon and
the engineering details I am trying to avoid).

Instead you are inventing a process that doesn't exist
and describing it in far more (and incorrect) detail
than exists in the published documentation.

As I have detailed above you are misunderstanding
what I
am saying

What you are saying is very different to what is being
done. It may be that this is because you are using terms
in an unconventional manner but you will then hit problems
in referring to your text books.

No. I am using terms and descriptions of mixers as in my
1985 Shrader Electronic Communication text which shows
a (tuned)resonant inductor and capacitor circuit for the
intermediate
and different ones for the sum and the input frequencies.
I dont want to keep arguing this point but what I am saying
is in principle what is being done. You are obscuring the essence
of what is being done which is the use of Fourier's transform to
obtain a Fourier series representation of the noisy received
oscillations
You also seem to have changed your understanding of the nasa
documents to come around to my initial impression


Again it may be clearer but it is wrong. It is not just
the carrier oscillations that are digitised, it is the
whole signal, oscillations plus random thermal noise
and any other sources such as the galactic background.

Your understanding is wrong. I did not say CARRIER
oscillations
You can't change the meaning of
'oscillations' to mean only the
part due to the spacecraft transmitter

"oscillations" means something regular,

Not necessarily. And obviously not in this context






I accept your apology but maybe you are similarly misreading
the nasa documents and that is why you are missing the
essence
of the procedure. You cant see the forest from the trees.

NASA don't talk of 'oscillations', they correctly talk of
the signal.

Nope, the amplitude of _all_ frequencies in the band is
calculated and passed on to the next stage without any
judgement.

I am talking about the final stage

The final stage is the carrier PLL, not the FFT. All
the FFTs are removed from the chain once the PLL locks
on and they play no further part in the process. It is
the PLL that tracks the drifting signal and gives us
the accurate measurement.

The FFT as I was using the term includes the PLL.

The two are entirely diffeent and separate.

Not the way I am using the term. Note I say how I am using the
term.

The point
which you insist on obscuring is that this technique
gets at the right sine frequency starting at the right time
from
the
sum of sine functions of various frequencies equivalent as
Fourier showed to
the noisy oscillations observed.

You said "an Fast Fourier Transform procedure is used to find
the underlying "sine" pattern of 1s and 0s that most closely
fits"

The FFT is not applied to "1s and 0s", it is applied to voltage
samples. The frequency is found and the PLL commanded to start
at
that frequency. The PLL locks on and tracks the carrier and it
uses a digital phase comparator that probably treats the signal
as 1s and 0s.

That is what I thought initially and you said I was wrong.
Evidently
you have changed your mind. It doesn't matter however for the
purposes of showing the essence of the procedure and the
rationale
as to why it is reliable.

There are several levels of processing that you are skipping
over
which are very important in establishing that the signal is
genuine and from the right craft. Ultimately that is your main
concern, isn't it?

Yes. I welcome your pointing this out. But I deplore
the obscure and argumentative way that you are doing it.


and I mentioned that the
movement of the Earth etc requires different patterns to
be
obtained successively but the point is that the FFT
procedure
finds the underlying pattern and it is this that is used
to
compare to the given sequence of 1s and 0s.

No it isn't. The final FFT is only used to set initial
frequency for the carrier PLL. If that locks, the
bandwidth is reduced to improve the signal/noise ratio.


You are saying the same thing that I was saying. I
think
it is clearer to say it without the jargon.

Clearer but completely wrong.
No clearer but not detailed.

The FFT does not compare
Again I did not say this. I said that after the
FFT procedure finds the dominant sine function,
this function is then compared to the observed set
of values which I thought you said earlier was reduced to
a set of 1s and 0s and that this was compared to
the corresponding observed set to get the degree of
error.


anything to a pattern of 1s and 0s. It does not compare
anything to anything else and in this case it does not
work on 1s and 0s.

The output of that is fed to the sub-carrier PLL.
Whatever the details a sequence of 1s and 0s
is obtained that is a digitised intermediate version of the
sky frequency.

No, a series of voltages samples like +0.25, -0.375, +0.112
etc.
is the result of digitising the IF.

Again
that has to lock before the signal can be decoded using
a phase detector. Then it gets decoded through the error
correction scheme. There are many critical steps after
the FFT, and in fact the FFT plays no part in the decoding
process whatsoever.

Again I did not say that it did.

The bottom line is a sine representation of a sum of sine
frequency
represention
of an oscillating pattern made possible by the FFT procedure
essentially
and this includes the phase locked loop procedure perhaps
involving
the recognition of some code modulation of the carrier to
insure that
the fitted frequency starts at the right time.


The fact that this representation is a much smaller
frequency than
the GHz sky frequency is ok because when you look at the
difference
between this and a small frequency representation of the
transmitted
frequency the difference is the same as the difference
between the
original frequencies. And it is this difference that is used
to
get the
Doppler shift.




This is the procedure I understood from your comments
and
various books and links.

You seemed to grasp it at the time, why
have you reverted to this grossly inaccurate
description
of the process?
Again I think you have misunderstood what I have
said. I dont think it is inaccurate if you replace single
intermediate
frequency by small range of frequencies around the single
intermediate frequency where small is relative the
original
frequency.


It is very inaccurate when the DSN document tells you
the analog band is the digitised band is 110MHz wide and
the signals of interest are of the order of 1Hz wide.

You are quoting the wrong document. We are talking about
the
intermediate frequency
being smaller that the original frequency. Is that so hard
for
you to understand.

What matters is how wide the frequency range is compared
to what you are looking at.

No what matters in this context is the size of the
intermediate frequency relative
to the size of the original frequency.



If the equipment only handles
a band that is small in comparison to the signal, the edges
will be chopped off, or if the Doppler shift was more than
expected the signal might be lost entirely. If th system
is 'wide band' then there is no such risk. How wide it is
compared to the original is completely irrelevant. Now this
matters because I know you are rferring to text boks and
those will use "wide band" and "narrow band" as terms relating
the width of the channle to the width of the signal, so if
you look up the text for "narrow band", you are going to get
entirely misleading information.

I am continually amazed that a person of your knowledge and
intelligence has so many
blindspots.

On the contrary, I can see potential mistakes you are
about to make through your unfamiliarity with the jargon
and I am trying to educate you in these terms to avoid
those pitfalls before you reach them.

And if you want to try and describe the digital
version of
the
mixer please do so. It was not clear from your emails.

The mixer is analog. The output is digitised and a baseband
extracted as shown on page 10. The details of the method
of mixing are not given but the principle is simply
multiplication of the incoming signal (including noise) by
the reference sine wave.

V_out = V_in * V_ref

where

V_ref = A * sin(wt)

This makes no sense. Electrical oscillations add by the
law of
superposition;

Yes, which is why it takes a special ciruit to get around that.

They dont multiply.

Dual gate fets and other methods of implementing mixers do

If they do then how do they. All I can see is superposition
and
then various filters to extract the desired frequency or range of
frequencies
Perhaps you have and analogue to digital converter to change
the incoming
frequencies to digital and then multiply them and then convert
this back instead
of the filter part of the mixer I see in my 1985 text.???


because that is their intended function and we poor designers
have to make them do it well. It's what engineers get paid
for (though I personally work on the digital side).

The mathematical fact that a sum of sine
and cosine functions can
be represented as a product of related sine and cosine
functions
has to be mentioned dont you think?

Only if you don't already know it.

Yes. And if I already knew it well we would not be having
this discussion would we?

The circuit multiplies
the two voltages together and since the product is the
same as a combination of sum and difference, you can then
discard all of (say) the sum components and keep all of the
difference components by a simple filter. Tuning is not
required, highly undesirable, and is definitely not included
in that part of the DSN system, it uses filtering instead
and to remove the jargon, that means it doesn't select a
single frequency from a range, it accepts the whole range,
treats it all equally, and only rejects a mirror image of
the range very far away.

You seem to have your own subjective
read of what others say and on what you say without understanding
that words are ambiguous and you have to say out loud what you
mean
or what you think is meant before going off halfcocked.
Ralph