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From: "Talbot Andrew" <ACTALBOT@dstl.gov.uk>
To: rsgb_lf_group@blacksheep.org
Subject: LF: RE: Phase meter?
Date: Wed, 9 Jan 2002 11:01:47 -0000
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>-
>From: boffin1 [mailto:boffin1@compuserve.com]
>Subject: LF: Phase meter?
>I have belatedly seen your queery about phase.  In the  abstract their is
>no answer to your question for phase reqires a reference i.e the phase of
>something relative to ....???     (i.e another part of the wave).   In the
>limit the phase must always close;  Nature abhors odd bits of phase 
>73, Roger,  G2AJV.
------------------------------------
The original question, I believe, was for software that would show the phase
of received signal / audio tone based on an internally generated reference.
This would obviously have to be at a specified frequency, but could easily
be generated by software using a Numerically Controlled Oscillator (a DDS in
software).  The procedure is to generate COS and SIN waveforms at the
specified frequency, which are jusst two sinewaves in software 90 degrees
apart.   Multiplying each of these by the incoming signal gives two
products, I and Q components respectively, these are usually then passed
through a  (software) low pass filter.  When displayed on a graph such that
the I compont is on the horizontal axis and the Q component on the vertical
a good representation of the signal in vector space is seen.  Such a display
is usually called a Vectorscope.

A clean tone with no noise appears as a dot on the screen, whose distance
from the origin is the amplitude.  A frequency error between the tone and
internal reference causes the dot to rotate around a circle at the
difference frequency - anticlockwise for a frequency too low and clockwise
for too high.  So this is a very good way of measuring exact frequency.
Noise appears as a amorphous circle around the dot.   The instantaneous
phase (relative to the internal reference, Roger) of a signal can be
determined immediately just by its position on the vectorscope

Soundcards are usually too unstable for this - their internal clock sources
are notoriously drifty and innacurate.  However, for those who understand
Windoze programming (All Hail !) it is quite a simple piece of software to
write.  It needs a user defined NCO which can be tuned in very fine steps -
0.01 Hz error only takes 100 seconds to complete a whole revolution.
Amplitude needs to be controlled - you can have a log/dB display of
amplitude versus radius. An alternative would be to use the left channel as
the signal input and the right as a user supplied reference tone.  This
would not have the versatility of an NCO appraoch, but could ensure absolute
freqeuncy stability of the display.

A more advanced and better solution would be to use a reference input tone
to continuously determine the Soundcard sampling rate, then use this
calculated figure in setting the NCO.

I use the 56002EVM module with its clock locked to a freqeuncy standard to
do the downconversion of a signal sampled at 8kHz.  The NCO is set in steps
of 8000 / 2^24 Hz.  The zero frequency I/Q samples are filtered and
decimated (reduced in sampling rate) to values of 800 right down to 1.95
samples per second.  These are output to a PC via a serial link for further
processing.  This basic tuning / decimation / filtering routine for the EVM
forms the workhorse of just about everything I do in the DSP line these
days, splitting the processing between two boxes really makes for easy
writing of software using a 66 MHz 486 running in a DOS environment !


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