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PICList Thread
2009\05\31@060611 by Peter Restall


Sean Breheny wrote:

> I suspect that your limiting factor may be the effect of pulsed noise
> on your high-Q filter. You might actually want to use a noise
> blanker-type circuit (detects spikes and prevents them from passing
> through into your narrowest filter, to prevent ringing).

Thanks - I'm not familiar with noise blanking so I'll have a look into that.
Switched capacitor filters have also been mentioned to me, but I've never
used those before either; possibly a more compact way than using LC or
op-amp filters, although they do have the disadvantage of requiring a clock
which may contribute noise into the system.

Olin Lathrop wrote:

> You're off by a factor of two and you're oversimplifying.  You make it sound
> like a L-C filter passband suddenly cuts off at a particular frequency.  In
> practise it has fairly steep sides, but the "width" of the passband depends
> on how much attenuation you need.

I should probably have qualified it with some sort of attenuation figure; it
did sound like a brick wall response now that you mention it.  The filter I
currently have on my breadboard has a -40dB bandwidth of about 10KHz according
to the software I used to design it; in practice it has fairly steep rolloff,
but difficult to measure with my equipment - my function generator doesn't
have a frequency read-out which makes it awkward.

> I've never read AN1115 nor do I plan to.  This is basic signal processing
> folks, with pretty simple math.  You'd be a lot better off understanding it
> than treating it like black magic describe in a sacred scroll.  There's no
> need to wave dead fish here.

AN1115 is far from a sacred scroll, but it does detail a method very similar
to what you originally described; so it was a logical document to include in
my further reading on the implementation details.

> They would, but then using a FIR is silly in the first place.  How did you
> get FIR filter in your head?  I certainly never said anything about using
> one.  I was envisioning a simple one or two pole low pass filter with each
> pole realized by:
>     FILT <-- FILT + FF(NEW - FILT)

AN1115 again !  The basic architecture they present is the sin/cos
multiplications followed by a FIR lowpass, decimation, and another FIR
lowpass.  FIR also seemed logical since they are (generally) linear phase
from what I understand.  It seems like I have missed something very simple

As a side note, I did get something working yesterday in Octave using the FIR
filters and successive decimations - 4 FIR filters with orders 12, 20, 30 and
40 (decimations of 3, 5 and 5).  Too much processing for a single ISR pass,
but I was looking at implementing it iteratively (as a state machine) which
should have left lots of cycles left over.  But that gets very complicated
and a nightmare to update should you change the filter characteristics.  It
also takes a fair amount of memory.  I'll revisit the drawing board looking
at simpler filters as you have said; it's not so easy to see when you are on
the wrong path if it is not one you have trodden before.

> I agree, but such a filter will need careful tuning.  Even then you have to
> think about drift over time and temperature.  That may be perfectly fine for
> a one-off hobby project if you're willing to tune it once a year and keep it
> at a constant temperature.

That's not the sort of solution I'm looking for - it should just sit there
and work, day-in-day-out.  I had something similar originally but it didn't
seem clean or reliable; there's something about using trimmers in such an
application that is decidedly unsatisfying.  This is one reason why I'm
looking into your DSP idea - it promises to keep the hardware simple at the
expense of more complicated (yet more flexible and updateable) firmware.  And
I get to learn lots of new and exciting things along the way, some of which
I'm hoping will sink in.


Pete Restall

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