Olin

Using your formula and simulating in Excel I always get an offset dependant
on FF  i.e. if using 1/16 then error is -15 counts and if 1/32 then error
is -31 counts.  Should this be added on at the end if using integer maths?

For example my line 11 is as follows:

=A10+INT(($D$1-A10)/$C$1)

Ian Forse

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> -----Original Message-----
> From: pic microcontroller discussion list
> [mailto:PICLIST@MITVMA.MIT.EDU]On Behalf Of Olin Lathrop
> Sent: 17 August 2002 23:08
> To: PICLIST@MITVMA.MIT.EDU
> Subject: Re: [PIC]: Real newbie question on ADC
>
>
> > This is fine if working in perfect conditions on a bench, but
> in the real
> > world, my circuits will be subjected to noisy industrial environments. I
> > have read elsewhere that multiple readings are required - sort of like
> > debouncing a pushbutton, I guess - but I have also read about 50Hz and
> 60Hz
> > hum.
>
> First, you have to characterize the signal you want and the noise
> you don't
> want.  What kind of bandwidth does your signal have?  How fast
> does anything
> need to respond to this signal?  Are you only worried about 50 or
> 60Hz power
> line noise or other random noise?
>
> Let's take the easy case where the desired signal has a much
> lower frequency
> than the dominant noise sources.  In that case, you take lots of readings
> and low pass filter the crap out of them.
>
> I often take one reading every 1mS, then do the rest in software.  The
> equation for the equivalent of an analog R/C low pass filter is:
>
>   FILT <-- FILT + (NEW - FILT) * FF
>
> where FILT is the current filtered value that is updated by this equation
> for each new reading, NEW is the new reading, and FF is the
> filter fraction.
> This is a number from 0 to 1 that controls what fraction of the
> input signal
> is "let thru" each iteration.  The filter time constant is a
> function of FF
> and how often this filter operation is performed.  An FF value of
> 0 results
> in an infinite time constant (the filter value never changes), and FF of 1
> results in a 0 time constant (output immediately follows the input).
> Obviously useful values are somewhere in between.
>
> The cool part comes from the observation that if you arrange FF
> to be 2**-N,
> where N is a positive integer, then the multiply by FF just becomes right
> shift by N.  You can see an example of this kind of filtering in the
> HAL_AD.ASPIC module at http://www.embedinc.com/pic/hal.htm.
> There the value
> of FF is defined by the assembler constant FILTB.  FILTB is actually the
> number of bits to shift right, which is 4 in this example.  Therefore FF =
> 1/16, although it is never computed nor used directly.  This
> filter yields a
> 50% step response in 11 samples, and a 90% step response in 36
> samples.  The
> attached SHR4.GIF show a plot of the step response.  If you only
> need about
> 100mS response to a changing input and you are taking a sample every 1mS,
> then you could use a smaller FF, like 1/32 or 1/64, for example.
>
> These filters can also be cascaded, just like you can have multiple R/C
> filters in series.  The attached file SHR44.GIF shows the step
> response of a
> two pole filter, each with a filter fraction of 1/16.  The step
> response is
> a little slower, but random input noise has been attenuated by an
> additional
> 12 dB for a total of 24dB.
>
> > I did post a question regarding the filtering of hum, and I was simply
> > pointed to some example code for filters - which was all a little
> daunting.
>
> Don't feel bad, there are entire graduate level courses that deal with
> filtering and signal processing.  There is a lot of hidden
> complexity, which
> may be why people prefer to just point at examples.
>
> > What is the trick? Is it just a matter of taking multiple readings at a
> > given frequency? I have read something about 40KHz - as it is a multiple
> of
> > both 50Hz and 60Hz
>
> 300Hz is also a multiple of both 50 and 60Hz.  Of course all multiples of
> 300Hz work too.
>
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