Manfred wrote:
> Hi all!
>
> Maybe someone here might help me with a DSP problem that should be
> trivial, but to me, as a newbie to DSP, is not.
>
> I'm developing a microcontroller-based speed controller for my small
> hydroelectric plant. I'm using a PIC 16F628 for this. I'm detecting the
> zero crossings of the 50Hz signal coming from the generator, and
measuring
> the duration of each half cycle, obtaining a signal at a rate of 100
> samples per second. The signal is in 16 bit format, and nominally 10000
> counts tall. Typical variations are a few tens of counts, and
occasionally
> as much as 1000 counts. I'm subtracting the 10000 reference, and using
the
> resulting error signal as input to a pro****tional-integral control
> function, whose output drives dump loads that burn off the excess output
> from the generator.
>
> The little problem I have is that a small 420Hz signal on the power
line,
> apparently coming from the generator's internal voltage regulator, is
> causing jitter in the zero crossings, which is in turn causing the
> pro****tional function of my controller to imprint an unwanted modulation
on
> its output. So I would like to apply a low pass filter function to my
> signal, with a cutoff frequency of roughly 20Hz, before the P-I
function.
>
> The question is: How can I implement this DSP low pass filter? It has to
> be in some simple way, because neither the PIC nor I can handle overly
> complex math!
>
> The only thing I can come up with is averaging the last several samples,
> perhaps with some weighing. But there must be some better method!
>
> It is desirable that the time delay in the filter be as short as
> possible.
>
> I would be most grateful for any help in this, be it with an explanation
> of how to implement the filter, or a hint as to where I can find this.
Manfred,
A simple low-pass filter might suffice. The conventional symbols used
for digital data streams are x[n] for the input data and y[n] for the
result, where n is the number of the sample in the sequence.
A unity-gain first-order filter is y[n] = k*x[n] + (1-k)*y[n-1]. It is
called an exponential averager; k is less than but close to 1. There is
usually an adequate choice of k that will allow the multiplications to
be accomplished with a ****ft and a subtraction.
I think your problem may be due in part to a (probably unwarranted)
assumption that there are no even harmonics in your generated waveform.
Even harmonics will cause alternate half cycles to have different
periods. You should measure the period of an entire cycle by triggering
on positive- or negative-going zero crossings, not both.
Jerry
--
Engineering is the art of making what you want from things you can get.
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