Dario Greggio wrote:
>> No, not with the half wave rectified method.
>
> ok, though I don't see an exact... reason. I mean, scaling it
> differently, it should give some good results all the same. But
> actually maybe it is no longer needed.

The method I thought you were referring to was finding the negative and
positive peaks, low pass filtering each of those seperately, then using the
difference between the two filter outputs to be the line voltage.  This
can't work now since the processor can't see the negative peaks.

> ok, my idea was to make this peak-measurement every 20mS, and average
> it
> with the previous one. Resimulated the peak method, and it fits in the
> 1.2% error indeed.

You are still sampling at 1KHz to find the peaks, right?  The 1.2% error I
mentioned comes from the 1KHz samples missing the peak by as much as 500uS.

> I also implemented this, storing 20 samples and grabbing the 3
> highest, and averaging them.

Hmm.  I'm not sure what that does to the worst case error.  You still don't
know where the peak was in time any better than before.  It may be that you
get a little less variation in the output as the sample frequency shifts in
phase with respect to the signal being sampled.  The constant to go from
this result to the input line voltage would also need to be a little higher
since you're never getting exactly the peak voltage.

This is where simulation is probably easier than going thru all the math.
Try a bunch of cases, each differing only in the phase between the 1KHz
samples and the input signal.  When using a single peak sample, the results
will be over a 1.2% range.  If you get a lower variation by averaging the
top 3 samples, then that sounds like a easy way to get a little less error.
The simulation can also tell you what the center scale factor is.  However,
I would worry about the effects of random noise on this method.

> Still have to try it on both simulation and
> hardware. What would you suggest, if you can point me to an
> "algorithm", in depth?

At this point I would probably apply a little low pass filtering to the raw
1KHz samples before doing anything more with them.  Two poles with a filter
fraction of .25 (2 bit shift) looks about right.  Every filter output sample
then is a function of a bunch of recent raw input samples, with random noise
attenuated by a factor of 16.  Let this do the "averaging" for you.  Now you
can pick off the peaks of the filtered signal and use them times a fixed
scale factor to get input line voltage.  Since each of these peaks will be a
function of a bunch of input samples, the result will have less variation
due to sample phase, in addition to having some noise immunity.

You got me curious, so I create a quick hack program to simulate this.  I
simulated one second plus one cycle.  You can see the result at
http://www.embedinc.com/temp/dario.csv.  The filters were still initializing
for the first cycle, so ignore that one.  I made the sample period 1.001 mS
so that the phase would change thru one whole cycle over the simulated
second.  You can see that the peaks of the filter output vary very little
over the full phase difference.  http://www.embedinc.com/temp/dario.gif
shows the cycle in the middle, when the input peak is straddeled evenly with
two sample.  Note that the top of the red curve doesn't quite reach 1.0.
That's the 1.2% error I was talking about before.

However, you'd be using the peaks of the cyan curve.  Those vary much less
with the phase because each of those samples is a function of a bunch of
recent raw input samples.  Look in the CSV file at the peak at 0.5085
seconds and the one at 1.009 seconds.  These are at the worst case opposites
of the phase.  The difference is less than 1/2 percent, so you can just
ignore it according to your ealier specs.

The program that generated this CSV file is at
http://www.embedinc.com/temp/dario.pas, although there is nothing really to
see in there.


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