From: Vladimir Vassilevsky on


Jerry Avins wrote:

> Vladimir Vassilevsky wrote:
>
>>
>> There is 1001 trick how to do harmonic analysis of truncated data.
>> However, for power industry purpose, the basic approach would probably
>> be good enough:
>>
>> 1. Sample 10 periods or more.
>> 2. Apply Blackman window.
>> 3. Then do the FFT.
>
>
> Agreed, but for Hi-Fi nuts, the accuracy won't be adequate if one
> samples 10.5 cycles.

BTW, just recently I had to measure ~0.0001% of THD. That's seismic data
acqusition specmanship. HiFi nuts, you said ? :-)


Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com
From: Jerry Avins on
Vladimir Vassilevsky wrote:
>
>
> Jerry Avins wrote:
>
>> Vladimir Vassilevsky wrote:
>>
>>>
>>> There is 1001 trick how to do harmonic analysis of truncated data.
>>> However, for power industry purpose, the basic approach would
>>> probably be good enough:
>>>
>>> 1. Sample 10 periods or more.
>>> 2. Apply Blackman window.
>>> 3. Then do the FFT.
>>
>>
>> Agreed, but for Hi-Fi nuts, the accuracy won't be adequate if one
>> samples 10.5 cycles.
>
> BTW, just recently I had to measure ~0.0001% of THD. That's seismic data
> acqusition specmanship. HiFi nuts, you said ? :-)

Can you discuss some of the subtleties involved? What kind of ADC is
linear enough to do that?

Jerry
--
"I view the progress of science as ... the slow erosion of the tendency
to dichotomize." --Barbara Smuts, U. Mich.
�����������������������������������������������������������������������
From: evilzucchini on
Thanks for the responses.

I don't think aliasing is a problem because our anti-aliasing filter's
cutoff is less than 2 times our sampling rate.

I am currently looking into window functions and trying to figure out a
good way to try and grab 10 cycles of data and do the analysis on that.

Thanks again,

Charlie
From: Jerry Avins on
evilzucchini wrote:
> Thanks for the responses.
>
> I don't think aliasing is a problem because our anti-aliasing filter's
> cutoff is less than 2 times our sampling rate.

I don't think aliasing is a problem either. If something I wrote gives
the impression that I do, please point it out to me.

> I am currently looking into window functions and trying to figure out a
> good way to try and grab 10 cycles of data and do the analysis on that.

A Blackman or Nuttall window ought to work well. Twenty cycles will do
better if you can afford the time. Starting and ending the sample train
with, say, positive-going zero crossings may work well enough so that
ten cycles do nicely. Remember not to include the tempting extra sample.
If a digitized cycle begins with a positive-going zero, it ends just
before the next positive-going zero.


Jerry
--
"I view the progress of science as ... the slow erosion of the tendency
to dichotomize." --Barbara Smuts, U. Mich.
�����������������������������������������������������������������������
From: glen herrmannsfeldt on
evilzucchini <cbarnes(a)digitalgridinc.com> wrote:
(snip)

> However, now our customers want more (big surprise). One of the requests
> was to do a Total Harmonic Distortion analysis of the currents (we are
> working in the power industry). I have no problem finding and
> understanding the mathematics for determining the total harmonic
> distortion, however, whenever I do it I get numbers that I believe to be
> very high. I am seeing 11-20% THD on "perfectly" smooth sine waves that I
> am generating. Perfect in the sense that I am adding in no harmonics and
> they look very clean on an oscilloscope. I am using Exocortex 2D FFT
> algorithm to generate the power of each harmonic on our sine wave.

There are different ways to say this. I usually say that the FFT
has periodic boundary conditions. That is, you get the transform
of the signal that is periodic with a period the length of the
transform.

> I suspect the problem is that I am not sampling perfectly in sync with the
> 60 Hz sine wave due to imperfections in the clock (like the rate I am
> sampling at would not be exactly 128 samples per cycle, but more like
> 128.5) and that this is making it look like there is more distortion than
> there is.

I agree.

> I did do a search trying to find the solution to this problem or at least
> some guidance, but I am not exactly sure what it is called so I have been
> unsuccessful trying to find a solution or even if I am barking up the right
> tree.

One solution is a PLL to lock your clock onto a multiple of the
input frequency. Traditionally, that takes hardware. With an FPGA
you might be able to do a DLL (digital locked loop). (Well, it
is still hardware, but you don't need to add any chips.)

The ability to do that, and I believe in general, is that the
signal not be do noisy that you can't determine the period.
For the PLL, that will be in the properties of the low pass
filter, and the time needed for it to lock. While noise might
be considered separately from distortion, it isn't always easy
to separate them.

-- glen
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