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PICList Thread
'Digital Filtering'
1996\10\28@115248 by J. Cabral

1997\03\20@012322 by David BALDWIN

> My project uses a PIC12C509 to monitor sensors and then outputs various
> simultaneous tones.  The tones are recorded on audio tape using a compact
> cassette recorder.

You may be able to save yourself some trouble if you record DTMF tones (if
that is enough) and then play it back through a DTMF decoder. You may be
able to find something on the web that will decode DTMF tones directly from
a .WAV file.

> Can anyone tell me the
> principles of software filtering or point me to web sites that may help?
I
> also need to know  about the file format of a WAV file. Can anyone help
> please?

For .WAV files; If you want lots of words, look for "Multimedia Programming
Interface and Data Specifications 1.0" by IBM and Microsoft. I think I got
it from Microsofts web site. If you want it in a simpler form, try
http://www.lightlink.com/tjweber/StripWav/Canon.html. It is a pretty simple
file format.

For learning DSP (now you're in the deep end), try
http://innet.com/~condor/wghome.html as a starting point and follow a few
links.

Steve.

======================================================
 Very funny Scotty.  Now beam down my clothes.
======================================================
Steve Baldwin                Electronic Product Design
TLA Microsystems Ltd         Microcontroller Specialists
PO Box 15-680                email: spam_OUTstevebTakeThisOuTkcbbs.gen.nz
New Lynn, Auckland           ph  +64 9 820-2221
New Zealand                  fax +64 9 820-1929
======================================================

1998\01\20@074232 by GERRY COX

1998\01\20@182807 by Mike Keitz

On Tue, 20 Jan 1998 12:21:10 -0000 GERRY COX <.....gcoxKILLspam@spam@DEK.COM> writes:
>The tape would be replayed and an accurate  reconstruction of the
>rates of
>acceleration and velocity of the shafts must be obtained.

Audio tape recorders don't preserve frequency and time very well,
especially recording on one machine and playing back on another.


>Perhaps there is a better way? I will have a look at the possibilities
>of
>PPM modulation though.

I think it would be best to record a digital code of the number of pulses
received on each input in a predefined interval of time controlled by the
recorder.  For example, the PIC would count for 0.1 second and record the
number ( 0 to 10) for each channel.  Using 6 bits to store the number of
pulses would take 60 bps for each channel, plus some overhead bits to
keep in sync and check for errors.

Even if the tape speed varies, exact rates and times could be recovered,
to the degree of accuracy of the clock of the PIC in the recorder.  Use
one of the modem standards and async data, and a standard PC modem could
be used to decode.  Some PC modems claim to support Bell 202 but details
are kind of uncertain.  If 300 bps is enough, use Bell 103.

At 13:23 20/03/98 +0000, you wrote:
>I wearily note that Web-browser/e-mailer software writers gladly spend time
> adding fancy features, instead of
>sorting out basic details. If they really want to impress, I'd like to see
> useful improvements like:
>
>* Detect and bounce binary attachments.
>* Binary attachments not copied when you press the "reply to" button
>* Detect and bounce messages from people trying to unsubscribe
>        by mailing to the PIClist instead of the list server.
>
>* Multi-priority stream and packet service channels.

Eudora (maybe Pro version only) gives good filtering - I can set it to
limit messages to any size, usually I leave it at 4kb. Later, when only the
first 4kb has been downloaded I can decide whether I want the lot or not.
It also has good filtering on any header content. I did decide to download
it all to see what it was - it then stitched all the individual parts back
together automatically and created the zip file.

Agent (which I only use for Usenet) has good filtering too in it's email
functionality I believe.

Regards

Adrian

WWW    WWW   Adrian Gothard
WWW WW WWW   White Horse Design
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'A newbie quesiton: What4s EMI filtering'
1998\07\12@235056 by David VanHorn

-----Original Message-----
From: Ricardo Ponte G <.....rniniverKILLspam.....CANTV.NET>
To: EraseMEPICLISTspam_OUTTakeThisOuTMITVMA.MIT.EDU <PICLISTspam_OUTMITVMA.MIT.EDU>
Date: Sunday, July 12, 1998 10:15 PM
Subject: A newbie quesiton: What4s EMI filtering


Hi to everyone:

       I4ve read somthing like: "EMI Filtering..."

       What4s EMI ?

Electro-Magnetic Interference.  AKA Noise, Hash.
It's made by your system, and you system can be glitched by EMI made
by other systems.



       Other things that I4ve read is: "Regular E-I and toroidal
transformer..."


       So, What4s E-I ?


One part of the core looks like an "E" and the other part looks like an "I"
The windings are wound on a bobbin, slipped over the "middle finger" of
the E, then the I is glued on top.  In iron xfmrs, the E and I laminates are
stacked alternately.  Take apart an old 6V power transformer and see for
yourself.

       What4s toroidal ?


Picture a ferrite donut (no strawberry filling!) with wires wrapped around
it.
It's harder to wind than the E-I type, but has less losses because it's all
one piece.

       If somebody Can help me, thanks.

We aims to please :)

'Median and Mean filtering'
1998\07\29@043415 by

Since getting my temperature controller to work, I have been playing around
with various filtering techniques to remove noise from my ADC output (AN512)

Lawrence kindly sent me the code to implement a median filter, which most
people seemed to think was the one to use.  Certainly, it rejects silly
values very well, however it doesn't seem to remove the high frequency noise
that a mean filter does.  The high frequency noise seems to be my biggest
headache at the minute, although mostly stemming from the use of an emulator
I think.

What I need is a way of combining the advantages of these two filters, but I
don't have a lot of RAM left so minimal overhead is needed.  Unfortunately
I'm having to deal with 16 bit values.  I currently use a circular buffer
with 8 entries and just use mean filtering.

Does anyone have any cunning ideas?

My only idea so far is to do a conversion and check that the conversion is
within a pre-defined range of the current average before putting it into the
buffer and recalculating the average.  Trouble is, any genuine large steps
would be ignored, such as at switch on.  Sounds like some kind of auto
tuning filter is needed maybe?

Regards

Mike Rigby-Jones
@spam@mrjonesKILLspamnortel.co.uk

1998\07\29@084930 by Chip Weller

Mike Rigby-Jones wrote:


|Since getting my temperature controller to work, I have been playing
around
|with various filtering techniques to remove noise from my ADC output
(AN512)
|
|Lawrence kindly sent me the code to implement a median filter, which
most
|people seemed to think was the one to use.  Certainly, it rejects silly
|values very well, however it doesn't seem to remove the high frequency
noise
|that a mean filter does.  The high frequency noise seems to be my
biggest
|headache at the minute, although mostly stemming from the use of an
emulator
|I think.
|
|What I need is a way of combining the advantages of these two filters,
but I
|don't have a lot of RAM left so minimal overhead is needed.
Unfortunately
|I'm having to deal with 16 bit values.  I currently use a circular
buffer
|with 8 entries and just use mean filtering.
|
|Does anyone have any cunning ideas?
|
|My only idea so far is to do a conversion and check that the conversion
is
|within a pre-defined range of the current average before putting it
into the
|buffer and recalculating the average.  Trouble is, any genuine large
steps
|would be ignored, such as at switch on.  Sounds like some kind of auto
|tuning filter is needed maybe?
|
|Regards
|
|Mike Rigby-Jones
|KILLspammrjonesKILLspamnortel.co.uk

Normally use of a median filter, or rejecting highest and lowest, is
used when there are very strange "shot" noise problems. If you are not
having data points which lay outside of 3 sigma of the mean it is
probably best to use all values.

I have had good luck using infinite impulse response filters (IIRF) when
filtering. You will want to oversample, but that is the case with any
filtering. Example (in C):
   FilterX16 -= FilterX16 >> 4;    // add 8 before shift to improve
rounding
   FilterX16 += NewValue;

This decays the filter value by 1/16th, which results in a time constant
(3db down, or 50% settling) of about 10.74 sample periods. Double pole
filters, which use two storage elements can be used to improved phase
delay (response time) while keeping the corner frequency at the same
location. Any text on digital signal processing should cover the design
of these filters. I typically design my own using a simple spread sheet
and graphing the response to a step and to some high frequency noise.
This allows me to tune using simple to handle multiplier factors. I
typically try to tune for minimum phase delays, which would correspond
to a Bessel filter.

By the way, the ADC method covered in AN512 doesn't strike me as
providing 16-bit result quality. If you are trying to create that sort
of resolution you probably are fooling yourself.

Chip.

1998\07\29@105437 by Mike Keitz

On Wed, 29 Jul 1998 09:25:07 +0100 "Rigby-Jones, Michael
[PAI01:4837:EXCH]" <RemoveMEmichael.rigby-jones.mrjonesTakeThisOuTNORTEL.CO.UK> writes:

>My only idea so far is to do a conversion and check that the
>conversion is
>within a pre-defined range of the current average before putting it
>into the
>buffer and recalculating the average.  Trouble is, any genuine large
>steps
>would be ignored, such as at switch on.  Sounds like some kind of auto
>tuning filter is needed maybe?


This is a good idea.  A little logic could be added to handle sudden
large changes in the input.  If most of the samples are being rejected,
the rejection process should be bypassed until the average becomes close
to the new value.  One simple technique would use a counter that counts
up when a sample is rejected and down if it is accepted.  If the counter
rises above a threshold, start putting all samples into the average, but
process everything else the same.  After enough new samples have been
averaged, the count will reduce below the threshold and the outlying
sample rejection will start working again.

Outlying samples have to be relatively rare in order for this to work, so
the new average will be close to the actual average.   If this isn't the
case, something could be done by increasing the rejection threshold to a
finite value, so even in the "bypass" mode, really bad samples are still
rejected.  Or take a single sample and force the average to it, then see
if the next few samples are close enough.


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1998\07\29@114657 by lilel

1998\07\29@144450 by Martin McCormick

1998\07\29@191955 by

Chip Weller wrote:

> By the way, the ADC method covered in AN512 doesn't strike me as
> providing 16-bit result quality. If you are trying to create that sort
> of resolution you probably are fooling yourself.
>
No, I don't think 16 bit resolution is realisticaly acheivable, the schmitt
trigger in the PIC was never designed for this use and I suspect dosen't
have terribly precise switching points.  However, I am getting around 9 1/2
bits over the temperture range of interest (15 to 45 degrees C) which is
more than adequate.  Thanks for the tips on digital filtering, it's
something which I would like to investigate further.

Mike Keitz wrote:

> This is a good idea.  A little logic could be added to handle sudden
> large changes in the input.  If most of the samples are being rejected,
> the rejection process should be bypassed until the average becomes close
> to the new value.  One simple technique would use a counter that counts
> up when a sample is rejected and down if it is accepted.  If the counter
> rises above a threshold, start putting all samples into the average, but
> process everything else the same.  After enough new samples have been
> averaged, the count will reduce below the threshold and the outlying
>       sample rejection will start working again.
>
Hmm..good idea, thanks Mike.  Given another few days of continuous bashing
of my head against the lab wall, I might have come up with this, you've
maybe saved some pain :)

Lawrence Lile wrote:
       The next thing to try might simply be analog filtering of your
       signal.  Nyquist will tell you that you have to sample at more than
       twice the maximum frequency of interest, and it is a good idea to
       make sure that no frequencies above that are present.  A couple of
       caps might just solve your problem adequately.   Often we are
       looking for signals that change over seconds, and measureing them at
       megahetz.  In this case a simple decoupling cap might be all that is
       needed.

The trouble is, the adc technique outlined in AN512 tends to preclude the
use of analogue filtering, adding a cap to the thermistor would just screw
up the timing.  You wouldn't have thought that the adc would be that prone
to noise as it has a reasonable sized cap (440nF in my case) straight to
ground on the sense wire.  Goes to show I don't know what I'm talking about
I guess!  I have to agree with your comments on emulators and noise though.
I guess it's time to get out the JW parts and see what's really going to be
happening.

For anyone using AN512 for tempertaure control, can I suggest that if the
system is a closed loop, then the thermistor is linearised, either
electronically or in software.  I used an interpolated look up table and the
loop stability was improved massively.  Not surprising when you look at the
thermistor characteristics.  The slope at the lower temperature range was
around 8 times that at the top end. Trying to tune the loop was a nightmare
until that was sorted.

Another hint, be carefull with emulators, the ICEPIC 2 can leave all the pic
periperals running when the core has actually stopped (selectable option).
Trouble is I didn't actually notice until my peltier cooler had reached
about 145 degrees C.  Strangely it seems to have lost a certain amount of
efficiency :o).

Regards

Mike Rigby-Jones
spamBeGonemrjonesspamBeGonenortel.co.uk

1998\07\30@031210 by Mal Goris

1998\07\30@050403 by STEENKAMP [M.ING E&E]

Idea:  If a message has "http://remove-list.com" in it, then it most
likely is UCE SPAM, and can be spam-binned <VBEG>

 Mark Willis, TakeThisOuTmwillisEraseMEspam_OUTnwlink.com

Hi, Bob,

Thank you for replying.

What I meant to say was that I could connect a capacitor(say 47u/17V)
between
the lines when it is off-hook. The actual location can be after the
switching
circuit inside the telephone set. I just want to know the simplest way
to mute
signals form the line while the DC connection is still active. In my
projet I
have to switch the earphone to monitor another signal.

Best reguards!


> From:         Bob Carter <RemoveMEwolftechTakeThisOuTMIDWEST.NET>
> Subject:      Re: How to filter out signals from Telephone lines
> Mime-Version: 1.0
> Content-Type: text/plain; charset="US-ASCII"
>
> A 16V cap across the phone lines?? The DC voltage is roughly 50V, and
an
> AC ring siganl can be over 100 volts. Can you place an approx. 20mA
load
> (to simulate off-hook condition) across your lines and then switch
in/out
{Quote hidden}

> your telephone with speaker?
>
> Thank You,
>
> Bob Carter, design engineer
>
> Wildlife Materials, Inc.
> 1031 Autumn Ridge Road
> Carbondale, IL 62901, USA
>
> voice (618) 549-6330 (ext. 225)
> fax   (618) 457-3340
> email wolftechEraseME.....midwest.net
> www   http://www.wildlifematerials.com

1998\11\09@204059 by Craig Webb

1998\11\09@213725 by Dennis Plunkett

Hi all,
   I have a PIC controlled CDI ignition circuit that I have been
building on for a while and I have got it down to all the voltages are
right but now when the PIC fires the SCR to discharge the CAP into the
coil the PIC seems to stop operating.

At first I believe the problem was that the I/O pins were being
rewritten so I re-initialized the I/O pins at different points in the
code and this allowed the PIC to operate longer but then it quit again
and I noticed that it had become very hot to the touch.

Does anyone have any ideas? Maybe some way to isolate the PIC from the
rest of the power. I am almost out of ideas. Anything will help.
Thanks
EraseMETroyignition.hell

1999\03\09@013747 by Tjaart van der Walt

"Troy P." wrote:
>
> Hi all,
>     I have a PIC controlled CDI ignition circuit that I have been
> building on for a while and I have got it down to all the voltages are
> right but now when the PIC fires the SCR to discharge the CAP into the
> coil the PIC seems to stop operating.
>
> At first I believe the problem was that the I/O pins were being
> rewritten so I re-initialized the I/O pins at different points in the
> code and this allowed the PIC to operate longer but then it quit again
> and I noticed that it had become very hot to the touch.

'Hot to the touch' sounds as if you are getting spikes on your
outputs that are running back into the PIC and shorting via
the protection diodes.

I have found nothing to be as fast as a tranzorb for absorbing
nasty spikes. Hook a 0.6Ke 5V tranzorb over the output pins.

--
Friendly Regards          /"\
                         \ /
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RemoveMEtjaartEraseMEEraseMEwasp.co.za  / \ AGAINST HTML MAIL
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|R&D Engineer : GSM peripheral services development|
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| Mobile : RemoveMEtjaartspam_OUTKILLspamsms.wasp.co.za  (160 text chars) |
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|--------------------------------------------------|

1999\03\09@024050 by Gerhard Fiedler

1999\03\09@173710 by Tony Nixon

Troy P. wrote:
> and I noticed that it had become very hot to the touch.

The 'spikes' may be causing the PIC to 'latch up' causing internal
shorts.

> Does anyone have any ideas? Maybe some way to isolate the PIC from the
> rest of the power.

My programmable ignition kit is now in over 2000 cars and have been in
operation for over two years.

I have a 100R resistor in series with each input to the pic that is
controlling any engine components, ie points, coil drivers etc. After
this resistor, on the PIC side, are a diode to 5V and one to GND. I used
IN5819's, but I have heard that ordinary IN4002's work OK. There is also
a 10n cap from the diode junctions to GND.

You may need to isolate the SCR from the PIC.

--
Best regards

Tony

PicNPoke - Multimedia 16F84 Beginners PIC Tools.

http://www.picnpoke.com
Email RemoveMEpicnpokeTakeThisOuTspamcdi.com.au

1999\03\10@113420 by Reginald Neale

'Filtering PIC Power maybe? Thanks'
1999\03\10@124525 by Troy P.

1999\03\12@172726 by Richard A. Smith

On Wed, 10 Mar 1999 09:34:45 +1100, Tony Nixon wrote:

>
>You may need to isolate the SCR from the PIC.
>

I just finished the first prototype of a unit that controls
a Cheston heater.  It's phase controled via 2 inverse
parallel SCR's which at peak run 350 amps at 480 VAC.  To
achive the proper isolation beetween the SCR's and the CPU
I had to use a gate trigger transformer.  The CPU sits on
the primary side and the SCR is on the secondary side
connected to the anode and the gate.  To fire the scr you
pulse the xformer.  This sends several large spikes into
the gate and turns on the SCR.  Be warned all gate xformers
are not created equal and you need to do a little research
to get right one for your SCR.  (Or at least in my case
they were)



--
Richard A. Smith                         Bitworks, Inc.
EraseMErsmithspamspamBeGonebitworks.com               501.521.3908
Sr. Design Engineer        http://www.bitworks.com

1999\03\12@200235 by Harold Hallikainen

On Fri, 12 Mar 1999 16:26:58 -0600 "Richard A. Smith"
<RemoveMErsmithKILLspambitworks.com> writes:
>On Wed, 10 Mar 1999 09:34:45 +1100, Tony Nixon wrote:
>
>>
>>You may need to isolate the SCR from the PIC.
>>
>
>I just finished the first prototype of a unit that controls
>a Cheston heater.  It's phase controled via 2 inverse
>parallel SCR's which at peak run 350 amps at 480 VAC.  To
>achive the proper isolation beetween the SCR's and the CPU
>I had to use a gate trigger transformer.  The CPU sits on
>the primary side and the SCR is on the secondary side
>connected to the anode and the gate.  To fire the scr you
>pulse the xformer.  This sends several large spikes into
>the gate and turns on the SCR.  Be warned all gate xformers
>are not created equal and you need to do a little research
>to get right one for your SCR.  (Or at least in my case
>they were)
>

       Might also try the MOC3010, MOC3020 series, etc.  These are
optically isolated triac drivers.

Harold


Harold Hallikainen
haroldSTOPspamspam_OUThallikainen.com
Hallikainen & Friends, Inc.
See the FCC Rules at http://hallikainen.com/FccRules and comments filed
in LPFM proceeding at http://hallikainen.com/lpfm

___________________________________________________________________
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1999\03\15@100419 by Richard A. Smith

On Fri, 12 Mar 1999 19:58:31 EST, Harold Hallikainen wrote:

>>the gate and turns on the SCR.  Be warned all gate xformers
>>are not created equal and you need to do a little research
>>to get right one for your SCR.  (Or at least in my case
>>they were)
>>
>
>        Might also try the MOC3010, MOC3020 series, etc.  These are
>optically isolated triac drivers.

Oh yeah I forgot those... for non phase control stufff
these will probally work great.  I had  trouble useing them
because I had to use the line voltage (480) to trigger the
SCR's.  To keep the gate from being destroyed at the peak
of the cycle I had to have a large resistor in series with
the gate.  But the voltage drop across the resistor kept me
from being able to trigger at < 30 or > 300 degrees. Which
in my application was important so I had to switch to
xformers.

IMHO the MOC's will be an eaiser (less complicated hadware
and software) soulution.


--
Richard A. Smith                         Bitworks, Inc.
spamBeGonersmithSTOPspamEraseMEbitworks.com               501.521.3908
Sr. Design Engineer        http://www.bitworks.com

On Mon, May 17, 1999 at 10:01:21AM -0700, Mark Willis wrote:
> Many admins are a little hard to convince to install "censoring"
> filters that censor by CONTENT - In the US at least, lots of people feel
> anything that might infringe on free speech is a BAD Thing, and you
> wouldn't want a wrongly set up filter [...]

At least in the US, one aspect of this is that, if you
*attempt* to filter posts, then you damned well better
do a good job.  For if you attempt to do something, and
you claim that you are doing that something, then there
is a good chance that you can be held liable if you fail
to do that something. Thus, if a list admin says that he's
filtering viruses, but one gets through, then a judge might
have to think long and hard before concluding that it was
not the admin's fault that someone's PC got a virus off
that list (this would probably hinge in part on whether
the admin took an appropriate level of care in this task,
with "appropriate" being to a large extent a function of
the admin's compensation -- or lack thereof -- for running
the list.)

If, OTOH, you do *not attempt* to filter viruses, and make
no claim that you are filtering viruses (or virus warnings)
or perhaps even *claim* that you make no attempt to filter
viruses or virus warnings, then it would be more clear
that the list admin is not responsible for the fact that
such are propigated via the list.

>   I 've learned to never downplay Virus/Worm warnings, they ALWAYS hurt
> someone far worse than I'd think at first.

In fact, in many cases, it seems that the virus warning
*is* the virus, with uneducated users acting as the vector.

--Bob

--
============================================================
Bob Drzyzgula                             It's not a problem
KILLspambobspamBeGonedrzyzgula.org                until something bad happens
============================================================

1999\05\17@143600 by Mark Willis

1999\08\25@141805 by Erik Reikes

1999\08\25@154332 by Dan Creagan

1999\08\25@161457 by Gordon Varney

1999\08\25@164403 by Adam Davis

1999\08\25@164412 by Larry Woods

1999\08\25@170043 by Dwornik, Sebastian

1999\08\25@175729 by Andy Kunz

>Sounds almost perfect to what I need, except for one thing. They state,
>"As the microwave radiation penetrates many materials, such as plastics
>and ceramics which are opaque to light, the sensor modules can for many

Have you considered optical (laser) ranging?

Two laser transmitters set to converge at a spot at the far end of your
expected range.

Use a CCD to accurately measure the distance between the dots.  This is
linearly proportional to the distance.

Simple geometry --> simple code.

Andy

==================================================================
Andy Kunz               Life is what we do to prepare for Eternity
------------------------------------------------------------------
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@spam@andy@spam@spam_OUTmontanadesign.com  http://www.montanadesign.com - Electronics
==================================================================

1999\08\25@183453 by eplus1

How do you know that the CCD is seeing a laser dot and not some other bright
dot of light?
Isn't a CCD kinda expensive?
Can you modulate the Lasers and use a photo diode with a filter with Q at
modulation frequency directed by a mirror on a pager motor? the filter
output pluses only when it sees the correct frequency and the time between
the two pluses is proportional to the distance.

James Newton, webmaster http://get.to/techref
(hint: you can add your own private info to the techref)
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{Original Message removed}

1999\08\25@184320 by Andy Kunz

>How do you know that the CCD is seeing a laser dot and not some other bright
>dot of light?

With a CCD, you can measure the dot.  If it's not the size of the laser
(within reason), you got the wrong dot.  Also, the color would be a good clue.

>Isn't a CCD kinda expensive?

They are pretty cheap any more.  Any small camera module (<$100) will do it.

>Can you modulate the Lasers and use a photo diode with a filter with Q at
>modulation frequency directed by a mirror on a pager motor? the filter
>output pluses only when it sees the correct frequency and the time between
>the two pluses is proportional to the distance.

Don't know - why not give it a try?

Andy

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1999\08\25@191457 by Peter van Hoof

1999\08\25@192740 by Dan Larson

1999\08\25@235341 by Tjaart van der Walt

{Quote hidden}

> Algorithm:
>         Pulse HIGH!     ; LED ON.
>         Read ADC        ; Result A = Signal + Noise
>
>         Pulse LOW!      ; LED OFF.
>         Read ADC        ; Result B = Noise
>
>         SUB A-B ; Result C = (Signal+Noise) - (Noise)
>
>         C = pure signal strength.
>
> This has proven to work well for distances up to 8 inches using an 8-bit
> ADC. This is the first time I'm trying to extend the range, with a more
> sensitive sensor and a 12-bit ADC.
>
> Is there a better algorithm of filtering which would provide greater
> results? A better sensor, or maybe a better method to the entire
> project. I have played with Sharps' GP2D02 module as well; and have
> gotten rather unsatisfactory results. If the current method cannot be
> improved upon, then my next step is to try one of those 40 khz
> Sharp IR modules. They won't give me proximity ranging, but
> maybe reliable object detection; and atleast that's a start.

What I would add is an AGC circuit. Do the switching faster
than the time constant of the AGC circuit, and you should
have more sensible readings.

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1999\08\26@082340 by Dwornik, Sebastian

1999\08\26@093109 by Gordon Varney

1999\08\26@093532 by Gordon Varney

1999\08\26@094746 by Gordon Varney

1999\08\26@110142 by eplus1

<BLOCKQUOTE AUTHOR="Andy Kunz">...Have you considered optical (laser)
ranging? Two laser transmitters set to converge at a spot at the far end of
your expected range.  Use a CCD to accurately measure the distance between
the dots.  This is linearly proportional to the distance. Simple
geometry --> simple code.
</BLOCKQUOTE>

<BLOCKQUOTE AUTHOR="Peter van Hoof">...no need for two lasers and two dots
if the camera is mounted a fixed distance from the laser...</BLOCKQUOTE>

<BLOCKQUOTE AUTHOR="Dan Larson"> ...you can turn the laser off, take a
frame, and if the dot is still there, it wasn't the laser.  Actually, you
could just take two frames. One with lasers on and the other with lasers
off.  Subtract them.  What's left are only the laser dots and some noise,
which could be
easily filtered.</BLOCKQUOTE>

I've been hearing this solution for a while and I'm sure it would work
nicely. It is further described at:
http://www.cyberg8t.com/pendragn/actlite.htm

The problem(s) is(are):
1. CCDs are in the under a $100 range. Ok for the dedicated hobbyist but it
makes it hard for kids, producible products, or wife/kid limited budgets
(me).
2. Reading a CCD and finding the right spot requires
  A. Speed. the standard 4Mhz PIC just ain't going to do it in real time.
Maybe an SX.
  B. Lots of RAM, especially to take two shots and compare. And the PIC
ain't got it.
3. It just seems to me (and I wish I had time to prove it) that its not
necessary. Again, modulate the Laser(s) and use a photo diode to a filter
with Q at modulation frequency directed by a mirror on a pager motor. The
filter outputs pluses only when it sees the correct frequency and the time
between the two pluses (or from the pulse to a reference provided by a
sensor that watches the mirror spin when using only one laser) is
proportional to the distance. Easily PICable, very low cost, etc...

Actually Scott Dattalo has done a brilliant page at
www.interstice.com/~sdattalo/technical/theory/dtmf.html
on how to pick out the presence a frequency by just sampling the signal at a
regular interval (ISR) and doing a DFT on it. He uses it for 0 hardware
DTMF. Scenix did it in their single chip modem. It could be used in a PIC to
eliminate the filter and just connect the photo-diode to the PIC. Another
pin for the reference pulse and maybe a PWM out for the motor speed.

Sigh... another for the "someday" bin.

James Newton, webmaster http://get.to/techref
(hint: you can add your own private info to the techref)
.....jamesnewtonRemoveMEgeocities.com
1-619-652-0593 phoneÊ



{Original Message removed}

1999\08\26@114154 by Andy Kunz

>1. CCDs are in the under a $100 range. Ok for the dedicated hobbyist but it

I thought it was for a commercial venture from the early part of the thread.

>   A. Speed. the standard 4Mhz PIC just ain't going to do it in real time.

We used a 20MHz PIC16C73 and had time to spare.  Used NTSC input to PIC.

>   B. Lots of RAM, especially to take two shots and compare. And the PIC
>ain't got it.

But you should already KNOW where the spot is (which lines) and only need
to scan them for which columns.  Using RLL encoding (how many pixels across
_didn't_ have the spot, when it appears, how many pixels does it occupy,
then how many don't again.  Three 1-byte entries per line.  Very little RAM.

Andy

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==================================================================

1999\08\26@125559 by Scott Dattalo

1999\08\26@132733 by eplus1

<BLOCKQUOTE AUTHOR="Scott Dattalo">It's one of those theory and practice
dilemmas... The only problem with this approach is that it only takes light
3ns to travel a meter. Now it is possible to resolve this kind of resolution
with fairly simple electronics. I could envision two square waves: one being
the square wave gating the laser on and off and the other being the
reflected signal. If you took the difference between these and ran it
through an integrator then 'theoretically' you'd have a dc signal
proportional to the phase difference. The phase difference is proportional
to the time-of-flight of the laser pulse.</BLOCKQUOTE>

Err... I think you may have misunderstood me. I was suggesting the use of a
rotating mirror to direct the attention of the photo diode across the plane
of the laser. For example, if the laser and mirror are mounted horizontally
from one another, the laser reflection will be somewhere on the horizontal
line scanned by the mirror. As the mirror turns, different points along that
line become visible to the photo diode. When the mirror reflects the point
that the laser "dot" is illuminating, the signal from the photo diode begins
oscillating at the frequency of the lasers modulation. This signal is
detected by a filter (analog or digital) and produces the pulse that is
detected by the processor. If the laser is on the left and the mirror on the
right, closer objects will cause the reflected "dot" to appear further to
the left from the viewpoint of the scanning mirror (pulse appears sooner)
and more distant objects will cause it to appear farther to the right (pulse
appears later) . The purpose of the modulation is only to quickly
distinguish the laser "dot" from other light sources as the mirror scans
across it.

The first picture at
http://www.cyberg8t.com/pendragn/actlite.htm
shows the mechanical setup except that I would replace the laser with the
photodiode so that different lasers could be used.

<BLOCKQUOTE AUTHOR="Andy Kunz">... NTSC input to PIC. ... already KNOW where
the spot is (which lines) and only need  to scan them for which columns.
Using RLL encoding (how many pixels across _didn't_ have the spot, when it
appears, how many pixels does it occupy, then how many don't again.  Three
1-byte entries per line.  Very little RAM. </BLOCKQUOTE>

Andy, you are absolutely clever... the CCD NTSC to PIC pin and count the
lines then the level is brilliant. If the laser modulation was genlocked to
the NTSC frame then you could look for the dot to be there then not be there
etc... just one bit to hold the expected value. The final data after 2
frames is just the depth of the point the laser is hitting. My only
objection left is the cost of the CCD. And the advantage of the CCD is that
any of the scan lines can be searched which would allow a laser with a beam
to line lens to produce an entire (vertical) 3d profile using only as much
memory as you wanted to store the results. Shifting that side to side
(slowly) would get you an entire 3d map of the objects in front.

James Newton, webmaster http://get.to/techref
(hint: you can add your own private info to the techref)
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1999\08\26@164534 by Anne Ogborn

1999\08\26@165406 by Anne Ogborn

1999\08\26@171506 by Dan Creagan

1999\08\26@172308 by Gordon Varney

1999\08\26@172915 by Andy Kunz

>Optical problem with this (and the CCD idea) is where to focuse
>the lense.

Sorry,  I assumed you all know that because the laser is intense, you can
cut the aperture _waaaay_ down.  This allows you to have a VERY WIDE focus
range.

Andy
==================================================================
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------------------------------------------------------------------
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==================================================================

1999\08\26@202533 by Anne Ogborn

1999\08\27@005011 by Tjaart van der Walt

Anne Ogborn wrote:
>
> Just sweep the laser -
> get a rotating mirror (lots of them around surplus).
> Put a photodiode in front of the laser as it starts it's sweep
> across, and another behind a lense focused on the scene ahead.
> When the laser reaches a given point, it will reflect off the
> scene into the receiver. The time difference is a one to one function
> of the range

I had an idea that I've been trying one of my buddies
(Hi Jules!) to try in his security R&D business.

It goes like this :
You suspend a light circular mirror from a stiff nylon fibre
with the reflective surface pointing down (dia = 3cm)
Stuck to the back of the mirror, there are three small magnets
(pointing up). Above the magnets, there are three small coils
with enough pull to deflect the mirror 45deg. Now you can scan
360deg. with a laser diode pointing upwards. Next to the diode
you have a light dep diode. As you scan the mirror, you record
the light intensity levels in, say 90 steps. These values are
saved in RAM, and the average calculated. The average is used
to normalise the saved values. This set of normalised, saved
values is compared to a pre-saved set of values (RMS sounds OK
for an indication), and any funny values will indicate an object
that shouldn't be there.

You could also use an AGC (Automatic Gain Control) circuit to
keep the measured values useable. For an AGC circuit, I think
the easiest would be to use a MOSFET like a VN10KM (Supertex)
in series with the Ri of an opamp. You need a non-inverting
amplifier for it to work. Stick the light level ouput of the
optical device into Ri. Use an integrated version of the output
to drive the MOSFET.

I probably didn't make any sense...

--
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|--------------------------------------------------|

1999\08\27@005026 by Tjaart van der Walt

"Dwornik, Sebastian" wrote:
>
> Can you elaborate on this "AGC" circuit? Any examples on the web would
> help.

I did in another post.

--
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|--------------------------------------------------|

1999\08\27@104058 by Dwornik, Sebastian

2000\01\21@114031 by Scott Dattalo

2000\01\21@124856 by Jeffrey Siegel

2000\01\21@170106 by Scott Dattalo

2000\01\21@214107 by Brian Aase

2000\01\24@054448 by Oleg Sergin

Hello Jeffrey,

May be all you need is math library. If all you want to remove from
signal is short peaks, exponential filter is enough. It has the
following transfer function
              1
      W(s)= ------
            Ts + 1
Calculation scheme is very simple.
Let us call x the result from ADC and filtered result.
Alpha is  ]0,1[ and determines time constant T. The bigger is alpha,
the smaller is T.
0) measurement ( x = ADC_result )
1) y = x
2) y_before = y
3) measurement ( x = ADC_result )
4) delta = x - y_before
5) y = y_before + alpha*delta ---> the next filtered number is ready
6) goto 3

So all you need is signed addition and multiplication. Be aware of
number of binary digits used to store y_before and to do all
calculation. If your ADC is 8-bit, I think you need 16-bit to get true
exponential filter. Your filter's step response shall look like this

|output
|
|1                 +++++++++++++++++++++++++
|            ++++++
|        ++++
|     +++
|   ++
|  +
| +
|+
+--------------------------------------------------- time
0

Sorry if I say trivial things :))

JS> Does anyone know of a software library (source code) that I could port to
JS> the PIC for real-time filtering?  I'm grabbing A/D data and want to do some
JS> basic artifact removal in software (real-time hopefully, before sending it
JS> to another processor) but I need a starting place of some working code.  Any
JS> language would be fine...

--
Best regards,
Oleg                             @spam@remove_this_word.serginspam_OUT.....fcita.dn.ua

2000\05\09@122608 by Andrew Kunz

2000\05\09@124441 by Alan B Pearce

2000\05\09@130321 by Alan B Pearce

'Data Filtering [OT]'
2000\05\09@134902 by Robert A. LaBudde

<x-flowed>At 12:24 PM 5/9/00 -0400, Andy wrote:
>I think what you want is a median filter.  Take multiple samples, sort them,
>then pick the one in the middle of the list.  Your spurious readings are
>extrema, right?

A more efficient, but just as stable, filter is a 10% or 20% trimmed mean:

E.g., collect a set of fixed size (e.g., 6-10), drop the high and low
result and average the remainder. With a set of size 8-10, you could also
drop the next highest and next lowest measurements.

The effective loss of information is only about 1/2 the number of points
dropped.

Obviously filtering of this type will cut the bandwidth by an equivalent
amount.

================================================================
Robert A. LaBudde, PhD, PAS, Dpl. ACAFS  e-mail: spamBeGoneralEraseMElcfltd.com
Least Cost Formulations, Ltd.                   URL: http://lcfltd.com/
824 Timberlake Drive                            Tel: 757-467-0954
Virginia Beach, VA 23464-3239                   Fax: 757-467-2947

"Vere scire est per causas scire"
================================================================

</x-flowed>

2000\05\09@150214 by Scott Dattalo

2000\05\10@020141 by Dr. Imre Bartfai

2000\05\10@042451 by Alan B Pearce

'Digital filtering on PIC's ?'
2000\05\11@230352 by KŸbek Tony

Hi,
I have an recursive moving average filter,
52 points with 3 bytes each, which works ok.
However the ram usage is 'a bit' high :-)
( almost the entire two top ram banks on my 16f876, )

Are there any other, more space conservatory, techniques
one could use to accomplish the same filtering ?

I do have 24 bit's of data for each sample, ofcource
not all bit's will change from one sample to next. But
was thinking of implementing an variable window ( 8 - 56 samples )
moving average filter to have an faster step-responce.
( big step = small window, little step = big window,
adjust automaticlly )

This however will not affect the ram usage, but also
was thinking about only saving the difference in successive
samples ( maybe 1.5 - 2 bytes ) and if difference is more
'reset' filter ( make the window very small ) until once
again the samples are in range. But think that this might
require quite abit of code to accomplish.

Good vs. bad ?

Anyone have any ideas ? or expirience with this ?


Thanks in advance,


/Tony










Tony KŸbek, Flintab AB            
ÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓ
E-mail: tony.kubekspamBeGoneflintab.com
ÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓ

2000\05\11@235440 by Plunkett, Dennis

2000\05\12@004749 by Andrew Warren

KŸbek Tony <RemoveMEPICLIST@spam@spamBeGoneMITVMA.MIT.EDU> wrote:

> I have an recursive moving average filter, 52 points with 3 bytes
> each, which works ok. However the ram usage is 'a bit' high :-) (
> almost the entire two top ram banks on my 16f876, )
>
> Are there any other, more space conservatory, techniques one could
> use to accomplish the same filtering ?

Tony:

The usual way to solve this problem is this:

   TEMP = AVERAGE * WIDTH
   TEMP = TEMP - AVERAGE
   TEMP = TEMP + NEW_SAMPLE
   AVERAGE = TEMP/WIDTH

If you do it that way, you need only enough storage for TEMP,
AVERAGE, and your NEW_SAMPLE.  If you set WIDTH to a power of 2, the
multiplication and division become trivial; if you pick a value of
256, the multiplications and divisions don't even require any code.

-Andy


=== Andrew Warren - .....fastfwd@spam@EraseMEix.netcom.com
=== Fast Forward Engineering - San Diego, California
=== http://www.geocities.com/SiliconValley/2499

2000\05\12@012229 by Erik Reikes

At 09:28 PM 5/11/00 -0700, you wrote:
>KŸbek Tony <.....PICLISTRemoveMEMITVMA.MIT.EDU> wrote:
>
>> I have an recursive moving average filter, 52 points with 3 bytes
>> each, which works ok. However the ram usage is 'a bit' high :-) (
>> almost the entire two top ram banks on my 16f876, )
>>
>> Are there any other, more space conservatory, techniques one could
>> use to accomplish the same filtering ?

It sounds like you have implemented an FIR filter with 52 taps.

What Andy has suggested is an IIR.  You can plug both of these into matlab
and compare the filter responses to see if the IIR is adequate.  

I forget the exact transfer functions for both, but any introductory DSP or
linear systems book will have them in there.



{Quote hidden}

>
>Tony:
>
>The usual way to solve this problem is this:
>
>    TEMP = AVERAGE * WIDTH
>    TEMP = TEMP - AVERAGE
>    TEMP = TEMP + NEW_SAMPLE
>    AVERAGE = TEMP/WIDTH
>
>If you do it that way, you need only enough storage for TEMP,
>AVERAGE, and your NEW_SAMPLE.  If you set WIDTH to a power of 2, the
>multiplication and division become trivial; if you pick a value of
>256, the multiplications and divisions don't even require any code.
>
>-Andy
>
>
>=== Andrew Warren - .....fastfwdSTOPspam@spam@ix.netcom.com
>=== Fast Forward Engineering - San Diego, California
>=== http://www.geocities.com/SiliconValley/2499

Erik Reikes
Senior Software Engineer
Xsilogy, Inc.

ereikesEraseME@spam@xsilogy.com
ph : (858) 535-5113
fax : (858) 535-5163
cell : (858) 663-1206

2000\05\12@042827 by KŸbek Tony

Thanks to all who responded, I will whip up Matlab
and se what effect will come of the FIR vs. IIR filters.
If it has the filtering/freq responce(damping) I need
then one could not comlain about ram usage :-) .


Dennis wrote:
>This depends on what the actual data is to represent and the
>procesing time that you have for each sample. If you have time, then
you

Data is from an AD7730 ( Transducer ADC ) measuring on strain gauges
in 2mV/V range. 5V AC to feed the brigde, will give me at most
10mV->24Bits
input signal. Aiming for, at least, 1uV verifiable resolution.
I want to cater for both high filtering application
where data output should be very stable but also applications where
a fast step-responce is crucial ( thats why the variable window idea ).
And yes I have quite a bit of 'free' time, I dont think data processing
time would pose as a problem.
( I'm more concerned over ram usage :-) )

All in all fun stuff...


/Tony






Tony KŸbek, Flintab AB            
ÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓ
E-mail: RemoveMEtony.kubekspamBeGoneflintab.com
ÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓÓ

'Digital filtering on PIC's ? [OT]'
2000\05\12@054421 by Robert A. LaBudde

<x-flowed>At 09:28 PM 5/11/00 -0700, Andy wrote:
>KŸbek Tony <spamBeGonePICLISTKILLspam@spam@MITVMA.MIT.EDU> wrote:
>
> > I have an recursive moving average filter, 52 points with 3 bytes
> > each, which works ok. However the ram usage is 'a bit' high :-) (
> > almost the entire two top ram banks on my 16f876, )
> >
> > Are there any other, more space conservatory, techniques one could
> > use to accomplish the same filtering ?
>
>Tony:
>
>The usual way to solve this problem is this:
>
>     TEMP = AVERAGE * WIDTH
>     TEMP = TEMP - AVERAGE
>     TEMP = TEMP + NEW_SAMPLE
>     AVERAGE = TEMP/WIDTH

The simplest method is to use an exponentially-weighted, moving average filter:

        NewEst = OldEst + alpha*(NewData-OldEst)

where
        OldEst = previous filtered estimate (start with first data point)
        NewEst = new filtered estimate
        NewData = current data point (measurement)
        alpha   = weighting factor

This is asymptotically equivalent to a moving average filter of run length
approximately 1/alpha. So alpha = 0.125 would roughly correspond to an
8-sample moving average, etc.

Notes:

1. Obviously there is a time lag between the filtered estimate and the
latest data (as there is in all integration filters) of about 1/(2 alpha)
samples.

2. If you are concerned about the increased error in the first few samples,
you can generate a slightly more complicated optimum (Kalman) filter. (This
involves a changing weighting factor).

3. The optimum value for alpha is about equal in magnitude to the 1-order
autocorrelation of the data.



================================================================
Robert A. LaBudde, PhD, PAS, Dpl. ACAFS  e-mail: ralspam_OUT@spam@lcfltd.com
Least Cost Formulations, Ltd.                   URL: http://lcfltd.com/
824 Timberlake Drive                            Tel: 757-467-0954
Virginia Beach, VA 23464-3239                   Fax: 757-467-2947

"Vere scire est per causas scire"
================================================================

</x-flowed>

2000\05\12@080546 by Morgan Olsson

2000\05\12@083505 by Scott Dattalo

On Fri, 12 May 2000, Robert A. LaBudde wrote:

{Quote hidden}

> At 09:28 PM 5/11/00 -0700, Andy wrote:
> >Kübek Tony <spamBeGonePICLIST@spam@MITVMA.MIT.EDU> wrote:
> >
> > > I have an recursive moving average filter, 52 points with 3 bytes
> > > each, which works ok. However the ram usage is 'a bit' high :-) (
> > > almost the entire two top ram banks on my 16f876, )
> > >
> > > Are there any other, more space conservatory, techniques one could
> > > use to accomplish the same filtering ?
> >
> >Tony:
> >
> >The usual way to solve this problem is this:
> >
> >     TEMP = AVERAGE * WIDTH
> >     TEMP = TEMP - AVERAGE
> >     TEMP = TEMP + NEW_SAMPLE
> >     AVERAGE = TEMP/WIDTH
>
> The simplest method is to use an exponentially-weighted, moving average filter:
>
>          NewEst = OldEst + alpha*(NewData-OldEst)
>
> where
>          OldEst = previous filtered estimate (start with first data point)
>          NewEst = new filtered estimate
>          NewData = current data point (measurement)
>          alpha   = weighting factor
>
> This is asymptotically equivalent to a moving average filter of run length
> approximately 1/alpha. So alpha = 0.125 would roughly correspond to an
> 8-sample moving average, etc.

Another way to view this formula is:

NewEst = (1-alpha)*OldEst + alpha*NewData

And taking Robert's alpha=1/8 example:

NewEst = (7/8) *OldEst + (1/8)*NewData
      =  ( 7*OldEst + 1*NewData) / 8

This formula is efficiently computed with this routine:

http://www.dattalo.com/technical/software/pic/twist.asm

I wrote it to adjust the relative wieghts of DTMF tones, but it works equally
well for these one-pole IIR filters. This version works for 8-bit values instead
of 24-bit values. However the efficiency concept is equally valid regardless of
the size of the data (it's the coefficients of the filter that are tweaked).

{Quote hidden}

>
> Notes:
>
> 1. Obviously there is a time lag between the filtered estimate and the
> latest data (as there is in all integration filters) of about 1/(2 alpha)
> samples.
>
> 2. If you are concerned about the increased error in the first few samples,
> you can generate a slightly more complicated optimum (Kalman) filter. (This
> involves a changing weighting factor).
>
> 3. The optimum value for alpha is about equal in magnitude to the 1-order
> autocorrelation of the data.

I posted an analysis of the frequency response of this filter back in Jan '00. I
don't have time to search the archives now. If you're going to perform any
matlab analysis this could be of use.

Scott

2000\05\12@092329 by Bennett, Matt

2000\05\12@203322 by picxpert

Ooh! Ooh! There's one available online for free:

http://www.dspguide.com/

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{Original Message removed}

2000\05\13@111046 by KŸbek Tony

>Curious: Why AC to the bridge?
>/Morgan

:-) , why not ?

Just kidding, the AD7730 can drive an brigde in AC-mode
where the excitation ( DC ) for the bridge is reversed
on alternate cycles hence the term 'AC'. This will 'elliminate'
thermocouple, offset and drift effects that one has in DC mode.
I.e. illiminate most DC errors.
Just a tad more complex system but much better performance.
( the only drawback is the settling time for the input after
the swithing, but even that is easily handled, for example
the AD7730 has an delay register to cater for this )
Optionally I will also drive the bridge with 10V ( +7.5 - -2.5 ) to get
more input.



/Tony



Tony KŸbek, Flintab AB            
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2000\05\13@111049 by KŸbek Tony

Hi again,

Scott Dattalo wrote:
>I posted an analysis of the frequency response of this filter back in
Jan '00. I
>don't have time to search the archives now. If you're going to perform
any
>matlab analysis this could be of use.

Cant seem to find it ? ( not in january atleast ).

Anyway my initial findings indicate that these filters are
, as stated by others, not equal. They have different freq.
responce and calculation properties.
The FIR I implemented is, albeit memory hungry, very fast
to calculate and have a linear step-responce.
( one additon and one subtraction/sample after initialisation )
I will look at the 'twist' routine by Scott and see what could
come out of it with some modifications to 24 bit, but I doubt
it will be more suitable than the original FIR.
I'm concered about the non-linear step-reponce in the IIR filter,
but it might not matter in the end.

But I'm by no means literate in filtering so I'm probably wrong :-).


Robert wrote:

>2. If you are concerned about the increased error in the first few
samples,
>you can generate a slightly more complicated optimum (Kalman) filter.
(This
>involves a changing weighting factor).

Do you mean that the weighing should leans towards the new sample
more at the start ? In that case that was my idea, I would also
use this in the 'running' filter to increase the step-repsonse.
( would have a similar effect as the variable window in the FIR )


>
>3. The optimum value for alpha is about equal in magnitude to the
1-order
>autocorrelation of the data.

Sorry, dont know if it's my english, but I dont really understand what
You mean here.


Nice to read all the comments though, never to late to learn.


/Tony





Tony KŸbek, Flintab AB            
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2000\05\13@133615 by Robert A. LaBudde

<x-flowed>At 01:25 PM 5/13/00 +0200, Tony wrote:
>Anyway my initial findings indicate that these filters are
>, as stated by others, not equal. They have different freq.
>responce and calculation properties.

True. You adjust those parameters to suit requirements.

>The FIR I implemented is, albeit memory hungry, very fast
>to calculate and have a linear step-responce.

The IIR EWMA formula I gave [ Y(n+1)=Y(n) + alpha*(X(n+1)-Y(n)) ] is about
as simple as it gets.

The formula is linear, so the filter is linear. I don't see the advantage
of a 'linear step-response'.

{Quote hidden}

> >2. If you are concerned about the increased error in the first few
>samples,
> >you can generate a slightly more complicated optimum (Kalman) filter.
>(This
> >involves a changing weighting factor).
>
>Do you mean that the weighing should leans towards the new sample
>more at the start ? In that case that was my idea, I would also
>use this in the 'running' filter to increase the step-repsonse.
>( would have a similar effect as the variable window in the FIR )

The Kalman filter is (without an error term) the exact average of all the
data so far experience, with a formula in recursive form.

The presence of a stochastic term de-tunes those exact formula to not
feedback all the information in the next same to the next filtered value.

The asymptotic form of the Kalman filter (after 100 samples or so for
transients to die out) is the EWMA form I gave above. The size of 'alpha'
is dictated by the relative size of the stochastic term. As I mentioned
before, the simplest interpretation is 1/alpha   ~ window length of a
moving average. In actuality, the weighting coefficients are alpha,
alpha^2, ..., a exponentially declining (geometric) series.

> >3. The optimum value for alpha is about equal in magnitude to the
>1-order
> >autocorrelation of the data.
>
>Sorry, dont know if it's my english, but I dont really understand what
>You mean here.

This is true only for small autocorrelation, in any event.

An 'exact' formula is:

autocovariance: C(k) = sum(k+1,n) {[ X(i)-Xave]*[X(i-k)-Xave]} /n

autocorrelation: r(k) = C(k)/C(0)

alpha = [1-sqrt( 1 - 4 r(1)*r(1) )]/(2*r(1))

This is 'invertible' (i.e., stable) for abs(r(1))<1/2.

================================================================
Robert A. LaBudde, PhD, PAS, Dpl. ACAFS  e-mail: EraseMEralRemoveMESTOPspamlcfltd.com
Least Cost Formulations, Ltd.                   URL: http://lcfltd.com/
824 Timberlake Drive                            Tel: 757-467-0954
Virginia Beach, VA 23464-3239                   Fax: 757-467-2947

"Vere scire est per causas scire"
================================================================

</x-flowed>

2000\05\14@133649 by KŸbek Tony

Robert wrote:

>The IIR EWMA formula I gave [ Y(n+1)=Y(n) + alpha*(X(n+1)-Y(n)) ] is
about
>as simple as it gets.

Yes, ofcource but as it uses multiplication ( even though
one could simplify with shifting data ) I meant that the
FIR with only addition and subtraction is 'lighter'.
And concerning PIC's in particular, to avoid rounding errors
one alomost is FORCED to use an 'alpha' that is 2,4,8,etc .
In the case of FIR the only time an 'uneven' window size matter
is during initialisation of the filter ( which is not required really ).
( I had an idea of gradually increasing/decreasing the window of
the FIR, but using the IIR i'm more or less forced to step up/down )

But anyway as the ram usage for this IIR filter is very low compared
to the FIR I will probably have both version and try them out in
'real life'.


>The presence of a stochastic term de-tunes those exact formula to not
>feedback all the information in the next same to the next filtered
value.
>The asymptotic form of the Kalman filter (after 100 samples or so for
>transients to die out) is the EWMA form I gave above. The size of
'alpha'
>is dictated by the relative size of the stochastic term. As I mentioned
>before, the simplest interpretation is 1/alpha   ~ window length of a
>moving average. In actuality, the weighting coefficients are alpha,
>alpha^2, ..., a exponentially declining (geometric) series.

Thanks, very clear and consice, I think I'm starting to understand
more of the implifications now.

/Tony


Tony KŸbek, Flintab AB            
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'Data Filtering'
2000\05\14@165722 by Alan B Pearce

'BIG NEWS! Server Topic Filtering [ADMIN]'
2000\05\16@130819 by jamesnewton

We (the list admins) have found a new feature in the list server software
that may provide a much better way of handling on and off topic (and sort of
off topic, etc..) posts. It would cause the list to work a bit differently
and might cause some confusion at first so we want to explain it and get
some feedback from the list on how best to implement it.

The new version of the list server can track "topics" by looking for
keywords or tags at the start of the subject line (after skipping past the
fw: or re:). It will then filter posts FOR the list members according to
which topics they wish to receive. For example, if we set the possible
topics to:
[PIC], [EE], [OT], and [AD]
we can set "topics of interest to subscribers" to [PIC] only and you will
only see posts with subject lines that start with "[PIC]:" (or "re: [PIC]:"
or "FW: [PIC]:" etc...). Of course you can send a simple command to the list
server that "turns on" the other topics as well. For example if you also
wanted to see [EE] posts (not directly related to the PIC, but still about
engineering) you could send:
spamBeGoneLISTSERV@spam@MITVMA.MIT.EDU?body=SET PICLIST TOPICS: +[EE]

Now, that is not the really good part.... here it is... <GRIN> if someone
DOESN'T put a topic tag at the start of the subject line, the list server
puts the post in a special topic called "OTHER" and if you haven't told the
listserver that you want "OTHER" posts, you wont get it! Cool huh?

The default settings for new subscribers will be with all topics including
OTHER turned ON so if they post without a topic, they will see their own
post. Also, we can set up an autoresponder that gets ONLY "OTHER" posts and
replies with a nice message that says "Um, you forgot to add [PIC]: [EE]:
[OT]: or [AD]: to your subject line, so, pretty much, I'm the only person
who saw your post, didn't you read the welcome message or the piclist.com
page?" followed by an explanation of what the topic tags mean, a mailto link
to turn off the OTHER topic (and the... other ([EE], [OT], [AD]) topics as
well), and a brief review of the list guidelines. And before anyone freaks
about looping on the autoresponder idea, we will put an [AUTOMATIC RESPONSE]
tag in and the autoresponder will know not to autorespond to posts
containing that.

Now, this is all a bit backwards from the way we are doing it now; rather
than asking that people be good and add [OT] to off topic posts, we will say
that all posts are not only off topic, but also completely ignored by
everyone but the autoresponder, unless you remember to put the topic at the
start of the subject line.

To get to the list and be read by all us members who are interested in
PIC's, a person will have to type [PIC]: at the start of the subject line.
This has the huge benefit of reminding people to think about whether or not
their post is really about PIC's. I personally think that this is an easy,
automatic and very complete solution to the problem of constant off topic
posting on the list. If it's really not about [PIC]:, you can add [EE]: or
[OT]: or [AD]: and it will go though to those people.

Using the same [OT], [EE] and [AD] strings for the topic names will allow
people who have set up automatic filters, sorters and classification rules
to continue using them. I know a lot of people move [OT]'s into a separate
folder for lower priority reading.

Of course, replying to an existing post will not change at all and people
may still take a [PIC]: related thread off into never, never land, but...
admins have to have something to do right? The no-HTML thing has been such a
big help that this list may get to be semi-self running and then we can
focus on other things like improving the FAQ or archives.

Ok, sorry for the long explanation, but we think this is an important new
direction for the list and will solve a lot of problems and make life a lot
nicer for members. What do you think?

RemoveMEjamesnewtonspam_OUTpiclist.com?subject=TOPICS_YES> jamesnewtonspampiclist.com?subject=TOPICS_YES_BUT <- please include
comments.
spam_OUTjamesnewtonspam_OUTspam_OUTpiclist.com?subject=TOPICS_NO <- please include why?

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2000\05\16@133124 by Andrew Kunz

2000\05\16@135025 by PIC development

2000\05\16@135821 by jamesnewton

Yes, the default for new (and old) members would be for all the topics to be
enabled. Most current members would turn off the OTHER: topic pretty quick.
Anyone who didn't want off topic discussions would turn off [OT]: Don't want
advertisements? Turn of [AD]:, Really not interested in non-PIC engineering
posts? Off goes [EE]: That leaves just [PIC]: as the minimum...

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-----Original Message-----
From: pic microcontroller discussion list
[PICLISTspamBeGone.....MITVMA.MIT.EDU]On Behalf Of Andrew Kunz
Sent: Tuesday, May 16, 2000 10:29
To: KILLspamPICLIST.....MITVMA.MIT.EDU
Subject: Re: BIG NEWS! Server Topic Filtering [ADMIN]


I personally hate having machines think for me.

I would hope the default for current members would be to enable everything,
and
only if they don't want it they can opt-out.

Andy

2000\05\16@140030 by jamesnewton

Correct, Actually it would tag it as being of topic OTHER: and send it only
to people who have OTHER: turned on.

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{Original Message removed}

2000\05\16@143946 by Jeff Frohwein

2000\05\16@170316 by Quentin

2000\05\16@172337 by Mark Willis

2000\05\16@172946 by jamesnewton

That is a very good point and is the one down point I can see in this.

Right now, if you want to post something about PICs to the PICList, you have
to:
A) Click on "New Message" or whatever button triggers your email client to
send email
B) Enter RemoveMEPICLISTRemoveMEEraseMEMITVMA.MIT.EDU in the "To..." field
C) Enter a subject line
D) Enter the post text in the body.

With the new system, step C would be modified to read:
C) Enter [PIC]: and the rest of the subject line.

Replying to a message would not be any different.

So... we are talking about requiring 6 extra keystrokes when posting. Its
not a lot, but it is extra work and I do understand that basically, this is
asking 99% of the PICList to do (a little) more so that the 1% that can't
bother to add [OT] reliably, will be reminded that they are on the subject
of PICs.

That's why we are asking for feedback on this idea. I still think its the
way to go, but I really will respect whatever the bulk of the list decides.
It may just not be worth it.

KILLspamjamesnewtonspamBeGonepiclist.com?subject=TOPICS_YES> jamesnewtonspampiclist.com?subject=TOPICS_YES_BUT <- please include
comments.
RemoveMEjamesnewtonspamBeGoneRemoveMEpiclist.com?subject=TOPICS_NO <- please include why?

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{Original Message removed}

2000\05\17@013537 by Josh Koffman

Notice however, that any posts marked [ADMIN] would go to the [OTHER]
autoresponder, and get a rude message back :) Just a thought...


Josh Koffman
@spam@joshySTOPspam@spam@mb.sympatico.ca

2000\05\17@161649 by Mark Willis

Awww, you spoiled my surprise for James <G>

 Mark

Josh Koffman wrote:
> Notice however, that any posts marked [ADMIN] would go to the [OTHER]
> autoresponder, and get a rude message back :) Just a thought...
>
> Josh Koffman
> joshyspamBeGonespamBeGonemb.sympatico.ca

'[ADMIN] Did the filtering scheme get abandoned?'
2000\05\30@122047 by Don B. Roadman

'All: [ADMIN] Did the filtering scheme get abandone'
2000\05\30@140435 by Mike Werner

'[ADMIN] Did the filtering scheme get abandoned?'
2000\05\30@140636 by Mark Willis

'[OT]: Digital filtering cont., some findings and i'
2000\05\31@043100 by KŸbek Tony

Hi,
Recently I asked the list of various techniques for
filtering ( in particular lowpass ). I received some excellent
replies and managed to implement most of them to try with
real-life data. My findings so-far is that, even though
fast step responce and high damping are normally, well,
opposite each other, I achived VERY good result by combining
an FIR with an IIR. I known to filter an filtered signal is
not really recommended but in this case, as the FIR
always are garantued to include all values within an specific range
( as the value just outside window is removed ) I though
that this should not impose any weird behaviour.
Anyway by using an FIR ( recursive average filter ) of fairly
small length 8,16 or 32 samples, feeding each new value into an
variable step IIR filter I get real life accuracy below 100 nV
( and this in an laboratory, i.e. all cables are loose
no ground-planes etc ). Measuring the delta between the FIR and current
FIR
I automaticly adjust the weighting of the IIR and receive an
very fast step responce when signal is variating and very high
dampening when signal is fairly stable.
The FIR's task in this case is ( mostly ) to remove low freq.
high amplitud disturbances ( for example an +/- 400 uV at 7 Hz )
of cource it will also filter out most noice ( +/- 200 nV ) also.

Are there something I'm missing here ? Can this setup cuase any
side effects ?


/Tony


Tony KŸbek, Flintab AB            
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E-mail: spamBeGonetony.kubekflintab.com
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I need to measure the frequency (or period) of a sinewave signal in the
range of 1KHz to 10KHz, which is only present for about 0.5 second.

The trouble is that there is another signal present ALL of the time that is
in the same frequency range and of approximately the same amplitude.  It
seems I need to filter out the always-there signal so I can measure the
intermittent one.  The exact frequency of the interfering signal is not
known, and could be higher or lower than the signal of interest.  It
probably must be determined before the filtering can occur, but that part
shouldn't be too difficult since the interfering signal is there prior to
the appearance of the signal of interest.

I realize that if the signal-of-interest happens to be the same as the
interfering signal, I probably cannot detect it, but if the filter is narrow
enough, I can live with that.

I thought I could do this with a couple of analog PLLs ahead of the PIC, but
lately I am wondering if I need something more along the lines of a FFT
system, or maybe a DSP chip.  I would really like to do everything in a
16F877 if possible.

If anyone could point me in the right direction I would appreciate it very
much.  Any and all suggestions welcome.

Thanks, Foster

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2001\07\09@164823 by Dan Michaels

Foster wrote:
>I need to measure the frequency (or period) of a sinewave signal in the
>range of 1KHz to 10KHz, which is only present for about 0.5 second.
>
>The trouble is that there is another signal present ALL of the time that is
>in the same frequency range and of approximately the same amplitude.  It
>seems I need to filter out the always-there signal so I can measure the
>intermittent one.  The exact frequency of the interfering signal is not
>known, and could be higher or lower than the signal of interest.  It
>probably must be determined before the filtering can occur, but that part
>shouldn't be too difficult since the interfering signal is there prior to
>the appearance of the signal of interest.
.........


Hi Foster, I have several ideas in mind - involving doing this in
software on a PIC. With an '877 running at 20 mhz, you can do A/D
sampling at 40 khz max. This would give you 4 samples/period with
your fastest signals, and should just about work.

1 - people have done FFTs on the PIC, but it looks like a lot of
   work, and you probably cannot get more than a few full calculations
   per second.

2 - since your interferring signal is always present, it looks like
   you could probably use an LMS adaptive filter, which would adapt
   to and reject the interference, but pass the target signal.

3 - if the signals are both sinewaves and the interfering signal is
   not doing any fancy phase jumps, it may be possible to simply
   lock onto the interfering signal with time sampling, and
   subtract it away from the composite target signal+interference
   to reveal the target signal alone.

With 20 mhz xtal and 40 khz A/D sampling, you have 200 intructions
to burn between samples, and you might be able to get away with doing
these kind of jobs.

However, since the PICs may not be fast enough to stay up with 40 khz
sampling rate, I was also thinking about the possibility of using a
Scenix, but those chips probably don't have enough RAM to do the job.

All in all, I think an adaptive filter, #2, might be something to
look at, short of going to a DSP chip.

best regards,
- dan michaels
http://www.oricomtech.com
========================

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2001\07\09@170444 by Olin Lathrop

> I need to measure the frequency (or period) of a sinewave signal in the
> range of 1KHz to 10KHz, which is only present for about 0.5 second.
>
> The trouble is that there is another signal present ALL of the time that
is
> in the same frequency range and of approximately the same amplitude.  It
> seems I need to filter out the always-there signal so I can measure the
> intermittent one.  The exact frequency of the interfering signal is not
> known, and could be higher or lower than the signal of interest.  It
> probably must be determined before the filtering can occur, but that part
> shouldn't be too difficult since the interfering signal is there prior to
> the appearance of the signal of interest.
>
> I thought I could do this with a couple of analog PLLs ahead of the PIC,
but
> lately I am wondering if I need something more along the lines of a FFT
> system, or maybe a DSP chip.  I would really like to do everything in a
> 16F877 if possible.

This is beyond a 16F877.  You probably want a DSP.  If you're very clever,
you might be able to squeak by with an 18 series PIC, but I wouldn't attempt
that unless you are pretty comfortable with digital signal processing
algorithms.  Unless this is a very high volume product where every cent of
parts cost counts, I would go for a DSP.  It's not going to cost that much
more and a decent 16 bit multiply/accumulate will probably be very handy.

As you mentioned, FFT is one possibility.  Another approach might be to do a
software PLL to find the noise signal, subtract it out, then look for the
occasional 500mS burst in the result.  Either way, there are some tricky
issues.  Go hire an expert unless you are very sure you're up to it.

How fast does the noise signal change frequency?  How often do the 500mS
burst of the signal to be measured occur?


********************************************************************
Olin Lathrop, embedded systems consultant in Littleton Massachusetts
(978) 742-9014, TakeThisOuTolinspamRemoveMEembedinc.com, http://www.embedinc.com

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2001\07\10@022123 by Scott Dattalo

On Mon, 9 Jul 2001, adastra wrote:

{Quote hidden}

> I need to measure the frequency (or period) of a sinewave signal in the
> range of 1KHz to 10KHz, which is only present for about 0.5 second.
>
> The trouble is that there is another signal present ALL of the time that is
> in the same frequency range and of approximately the same amplitude.  It
> seems I need to filter out the always-there signal so I can measure the
> intermittent one.  The exact frequency of the interfering signal is not
> known, and could be higher or lower than the signal of interest.  It
> probably must be determined before the filtering can occur, but that part
> shouldn't be too difficult since the interfering signal is there prior to
> the appearance of the signal of interest.
>
> I realize that if the signal-of-interest happens to be the same as the
> interfering signal, I probably cannot detect it, but if the filter is narrow
> enough, I can live with that.
>
> I thought I could do this with a couple of analog PLLs ahead of the PIC, but
> lately I am wondering if I need something more along the lines of a FFT
> system, or maybe a DSP chip.  I would really like to do everything in a
> 16F877 if possible.
>
> If anyone could point me in the right direction I would appreciate it very
> much.  Any and all suggestions welcome.

Foster,

You didn't explicitly say it, but I'm assuming the "noise" is a sine
wave too. If so here are a couple of suggestions:

1) Digital phase lock loop

With the DPLL, you can lock onto the "noise" sine wave. As long as the
DPLL is locked, there should be little or no correction applied.
Suppose the loop filter has a very long time constant. It may take a
long time to lock, but it will tend to also stay locked or even
continue to "run" even if the reference is removed.

Use the error signal in this DPLL as an indicator for when the "noise"
is being perturbed by your signal. Let this start a second DPLL that
has a much shorter time constant.

Now if you're really clever, you could use the first DPLL to cancel
out the "noise" sine wave!

2) autocorrelation filter.

Suppose your unknown signal is not present.  If so, then you should be
able to take a sample of the "noise" and see that it's correlated to
itself. The easiest way is perhaps to do something like:


f(t+i*dt) = samples at t, t+dt, t+2dt, ... t+Ndt

I = sum( f(t+i*dt)  *  f(t+i*dt + M*T)) / sum(f(t+i*dt))
Q = sum( f(t+i*dt)  *  f(t+i*dt + (M+1/4)*T)) / sum(f(t+i*dt))

Where
M is an integer
T is the period of the known noise (determined by another algorithm)
dt is the sampling interval
N is the number of samples
I is the "in phase" component
Q is the "out of phase"

In other words, a sample window of the signal is multiplied with a
copy of itself that has been shifted an integer # of cycles and the
sum of products is obtained. Similarly, a second window is shifted an
integer # plus an extra quarter to obtain the "quadrature" component.
(The stuff in the denominator of the two expressions normalizes the
result.)

Strictly speaking, this is NOT autocorrelation. Autocorrelation would
compute an infinite number of the above computations and not just two.
In other words, the window over which the signal slides varies
continuously and not just M*T. However, the autocorrelation of a sine
wave produces a series of delta functions spread M*T units apart. So
if you can zero in on one of the delta functions you'd tend to save a
little computation time!

Now, the goal, like in the DPLL, is to determine when the
autocorrelation peak is perturbed. Again, like above,  you could
control the rate at which the window is adjusted. So sudden changes in
the input would manifest them selves in reducing the autocorrelation.
This information can be used as a trigger to begin looking for the
other signal.

One of the problems with the autocorrelation is that it is incapable
of distinguishing harmonics. In other words, if the noise is a 5kHz
sine wave and the signal is a 1kHz sine wave, the autocorrelation
would not change.

Another problem is that the autocorrelation approach is
computationally expensive.

3) Filter banks

Since we're talking about computationally expensive algorithms, we
might as well mention filter banks. In the brute force case, you can
imagine having say 10 notch filters centered on 1kHz, 2kHz, etc. and
each having a bandwidth of around 2kHz. You could monitor the output
of each of these filters and then look for changes in signal strength
output. When a change is detected, then that's a clue to start looking
more closely in another area.

A brute-force filter bank is a DFT. It will take the time domain
signal and place the energies of each frequency into the appropriate
bins. However, unless this information is going to be viewed by a
human, I'd argue that it's a total waste of resources. For one reason,
it computes all of the frequency decomposition up front - even for
those frequency band for which it's not needed. Second, a DFT provides
no way to control the width of the frequency bins.

Having a set of discrete filters solves both problems mentioned with
the DFT. In addition, a set of discrete filters requires fewer RAM
resources. However, this approach is not computationally efficient.
The DFFT is of order N*ln(N) and the filter bank is M*N where M is
the number filter banks. A bank of just 8 filters is about
computationally inefficient as a 256 point DFFT. However, instead of
discrete filter banks it's possible to construct cascaded filters
constructed from "Quadrature Mirror Filters". For example, suppose you
split the frequency band into two parts: LOW and HIGH. The LOW portion
is passed through another filter that splits it again into low and
high portions. Similarly the HIGH portion is split in two. This will
result in the original signal being split into four pieces. Now the
trick is that since the filters are cascaded there's certain
information known. For example, when the LOW portion is generated,
there are no frequencies from the "HIGH" portion. Consequently the
samples fed to the filters cascaded off the LOW portion can be
"decimated", i.e. the sample stream can be sub-sampled.

(This whole subject is heavy DSP {and beyond the scope of a [PIC]:
tag} and discussed in "Advanced Digital Signal Processing - Theory and
Applications", Zelniker and Taylor, ISBN 0-8247-9145-2. In fact, a
problem quite similar to this is discussed in the section that
introduces Wavelet Transforms.)

The filter bank approach won't find your signal per se. However, it
can be a relatively quick way of determing that the signal is present.
I imagine this couple with a DPLL would be the best solution. If
you're stuck with the '877 then there's always the "1-bit" tone
decoder that can be used to put the signal into frequency bins:

http://www.dattalo.com/technical/theory/dtmf.html

Scott

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2001\07\10@095743 by Douglas Butler

2001\07\10@141353 by Dal Wheeler

Well, I suppose it sort of depends on how fast the constant signal changes,
but...  I wonder if one could sample the continuous signal and synthesize an
equivilant waveform; continue the synthesized waveform based on the measured
signal just prior to the window of interest.  Then you might just subtract
the synthesized signal from the other.  O'Course I havent tried anything
like that so I might just be talking out my backside...

-Dal
----- Original Message -----
From: Douglas Butler <EraseMEdbutlerSTOPspamRemoveMEIMETRIX.COM>
To: <spam_OUTPICLISTRemoveMEEraseMEMITVMA.MIT.EDU>
Sent: Tuesday, July 10, 2001 7:50 AM
Subject: Re: [PIC]: Suggest an approach to this filtering problem?


> If you could synchronize your sampling to the zero crossings of the
> interfering signal it would disappear.  I think filtering in the time
> domain will be tough.
>
> Sherpa Doug
>
> > {Original Message removed}

2001\07\10@153934 by Dal Wheeler

Along those lines of thought, since we're just interested in the
frequencies, why not stick a zero crossing detector and count pulses with
the pic against a timer to get a frequency.  Subtract the current pulse
block count from that of .5 seconds ago, ifany significant increase from
zero --there's your intermittant signal freq. Again, your continuous signal
can't change significantly fast or this'd frag.

!???
-Dal
----- Original Message -----
From: Dal Wheeler <TakeThisOuTdwheelerRemoveME@spam@insightek.com>
To: pic microcontroller discussion list <EraseMEPICLISTRemoveMEMITVMA.MIT.EDU>
Sent: Tuesday, July 10, 2001 12:09 PM
Subject: Re: [PIC]: Suggest an approach to this filtering problem?


> Well, I suppose it sort of depends on how fast the constant signal
changes,
> but...  I wonder if one could sample the continuous signal and synthesize
an
> equivilant waveform; continue the synthesized waveform based on the
measured
{Quote hidden}

> signal just prior to the window of interest.  Then you might just subtract
> the synthesized signal from the other.  O'Course I havent tried anything
> like that so I might just be talking out my backside...
>
> -Dal
> ----- Original Message -----
> From: Douglas Butler <spamdbutler.....spamIMETRIX.COM>
> To: <PICLISTspam_OUT@spam@MITVMA.MIT.EDU>
> Sent: Tuesday, July 10, 2001 7:50 AM
> Subject: Re: [PIC]: Suggest an approach to this filtering problem?
>
>
> > If you could synchronize your sampling to the zero crossings of the
> > interfering signal it would disappear.  I think filtering in the time
> > domain will be tough.
> >
> > Sherpa Doug
> >

> > > {Original Message removed}

2001\07\11@122108 by adastra

Hello for all.

I'm looking for way to do
60Hz BAND-PASS FILTER and
60Hz NOTCH FILTER using a PIC.  
Is it possible to do this with a PIC?  
Have you ever seen something like this?  
Thanks in advanced goes any help    Luís Fernando

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2001\08\13@082734 by Ing. Gustavo Quilosa

Yeap ... it's possible....
.... try with the IIR filters algorithms of the embedded handbook.

----------------------------------------------------------
Ing. G. Quilosa
Development Dept.
----------------------------------------------------------

----- Mensaje original -----
De: antoniasse <antoniasseKILLspamEraseMEBRAILE.COM.BR>
Para: <EraseMEPICLIST@spam@@spam@MITVMA.MIT.EDU>
Enviado: lunes 13 de agosto de 2001 9:01
Asunto: [PIC]:Digital filtering using a PIC


Hello for all.

I'm looking for way to do
60Hz BAND-PASS FILTER and
60Hz NOTCH FILTER using a PIC.

Is it possible to do this with a PIC?

Have you ever seen something like this?

Thanks in advanced goes any help

Lums Fernando

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2001\08\13@121421 by Harald Milz

In article <001f01c123f4$6bb91a90$0101a8c0@desarrollo01>, Ing. Gustavo Quilosa <RemoveMEquilosabeKILLspamRemoveMEyahoo.com> wrote:
> Yeap ... it's possible....
> .... try with the IIR filters algorithms of the embedded handbook.

Doesn't that depend on the signal's sample rate?

> De: antoniasse <TakeThisOuTantoniasseBRAILE.COM.BR>

> I'm looking for way to do
> 60Hz BAND-PASS FILTER and
> 60Hz NOTCH FILTER using a PIC.

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2001\08\13@131318 by t F. Touchton

part 1 1641 bytes content-type:text/plain; charset=us-ascii
for 60Hz notch... sample at an even harmonic frequency and average the
result.

Scott F. Touchton
1550 Engineering Manager
JDS Uniphase



                   Harald Milz
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In article <001f01c123f4$6bb91a90$0101a8c0@desarrollo01>, Ing. Gustavo
Quilosa <quilosabeSTOPspamyahoo.com> wrote:
> Yeap ... it's possible....
> .... try with the IIR filters algorithms of the embedded handbook.

Doesn't that depend on the signal's sample rate?

> De: antoniasse <antoniasseSTOPspamKILLspamBRAILE.COM.BR>

> I'm looking for way to do
> 60Hz BAND-PASS FILTER and
> 60Hz NOTCH FILTER using a PIC.

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what
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part 2 3178 bytes content-type:application/octet-stream; (decode)

part 3 136 bytes
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2001\08\13@133616 by Saurabh Sinha

Hey Luis,

We did somehting similar, you can have a look at my webpage. Look under
filters, there's some files that may help (FIR implemenetaion, but should
give an idea).

(Though, I must say I dont think you really need to do this on a PIC!)

Regards,

S. Sinha (Saurabh)
----------------------------------------------------------------------------
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Tel: (012) 343-4912
Cell: 0825405577
Visit: http://home.mweb.co.za/si/sinha ['H']
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{Original Message removed}

2001\08\13@143143 by Scott Dattalo

On Mon, 13 Aug 2001, Scott F. Touchton wrote:

>
> for 60Hz notch... sample at an even harmonic frequency and average the
> result.
>

With the added benefit that you'll notch 30Hz, 15Hz, 7.5Hz, ...
and in general 60/N Hz, including when N -> inifinity. In other words,
you'll filter DC too!

I'm not sure of the application, but you'll probably want to filter
60Hz by integrating over 1/60'th of a second time interval. For
example, if you had a voltage to frequency converter as your A/D then
you can count the pulses it emits for 16.6666 mSec. Or if you have a
successive approximation A2D, then you could synchronously sample say
60*16 times per second and average the last 16 samples (add them
together and then shift right 4) to get a representation of the
signal minus 60Hz. Note that both of these suggestions are really low
pass filters with a notch at 60Hz and not Notch filters per-se.

Scott

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2001\08\13@213330 by Tom Mariner

2001\08\14@073459 by Vasile Surducan

Sinha, you have a very nice theory here !  Just downloaded.
Cheers, Vasile


On Mon, 13 Aug 2001, Saurabh Sinha wrote:

{Quote hidden}

> Hey Luis,
>
> We did somehting similar, you can have a look at my webpage. Look under
> filters, there's some files that may help (FIR implemenetaion, but should
> give an idea).
>
> (Though, I must say I dont think you really need to do this on a PIC!)
>
> Regards,
>
> S. Sinha (Saurabh)
> ----------------------------------------------------------------------------
> ----
> Tel: (012) 343-4912
> Cell: 0825405577
> Visit: http://home.mweb.co.za/si/sinha ['H']
> Visit: http://ieee.up.ac.za/members/saurabhs ['W']
> Visit: http://www.activesms.co.za/?sponsor=270825405577 [ActiveSMS, join
> me!]
> ICQ: 82010326
> Alternative e.mail: .....ssinha.....RemoveMEmweb.co.za (if the auto-reply address doesn't
> work)
>

> {Original Message removed}

2002\02\28@110346 by Michael Vinson

2002\02\28@112224 by michael brown

2002\02\28@113420 by Michael Vinson

2002\02\28@113621 by 859-1?Q?Alexandre_Guimar=E3es?=

2002\02\28@114659 by Lawrence Lile

2002\02\28@115947 by Robert A. LaBudde

At 10:18 AM 2/28/02 -0600, Michael wrote:
> > michael brown wrote:
> > >This may sound like a dumb question, but I have not run across a good
> > >"plain
> > >English" description of low-pass [...]
> >
> > A time-varying signal can be represented as a function of time, f(t).
> > The same function can be represented in "frequency space" via a linear
><SNIP>
> >
> > Hm, this isn't really "plain English" is it? Try again.
>
>Not to me, it pretty much reads like the rest of the "explanations" I've
>seen.  ;-)

A low-pass filter is simply a moving-average in a sampled data system.

An analog system is continuous, so you have to visualize an infinite number
of infinitesimal pieces going into the moving-average. In this case, the
moving-average is called an "integral" and the process of adding up the
infinite number of pieces is called "integration".

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2002\02\28@124851 by Sean H. Breheny

2002\02\28@181013 by Olin Lathrop

> It reinforces what I already understand about
> filtering electrical or electro-magnetic "waves".  I am looking for the
> definition as it applies to EE and sampled data.

Inside a computer a simple low pass filter working on successive samples is:

Filt <-- Filt + (New - Filt)*FFrac

Where FILT is the filter that is being updated, and NEW is the new sample
being accumulated into the filter.  FFRAC is the filter fraction, a value
from 0 to 1 for a normal low pass filter.  It governs how "heavy" the
filtering is.  A value of 0 is infinitely heavy, which means the filter
output never changes at all.  A value of 1 causes the input signal to be
passed thru without any filtering.  Do an example on paper to see how this
works.  I often cascade two of these things, called "poles", to make a "two
pole" digital filter.


********************************************************************
Olin Lathrop, embedded systems consultant in Littleton Massachusetts
(978) 742-9014, EraseMEolinspamBeGoneKILLspamembedinc.com, http://www.embedinc.com

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2002\02\28@181518 by michael brown

2002\02\28@221959 by michael brown

I already know that you know this Michael.  The way that you probally know it
the best would be the simple RC (resistor-capacitor) network.  A typical RC
network would be one where the resistor is in series with your signal and the
capacitor is to ground.  The "physically related" process that you seek is an
understanding of the capacitor and the rate that it charges and discharges.

Example #1:  You have a large diameter garden hose that you are turning on and
off at a slow rate.  When the garden hose is on, water goes out. When off, water
is sucked back up into the hose.  Put the garden hose into a small bucket and
have it turn on and off once per minute.  You can EASILY watch the level of the
water rise and fall within the bucket.

Example #2:  Small garden hose inside a large bucket and you are turning the
garden hose on and off at a high rate (lets say 20 times per second).  You would
not notice the water rise and fall very much if at all.  Mostly due to the small
amount of water going in and out of the small bucket.

Apply this to the real world:  The larger the size of your resistor, the smaller
the diameter of the garden hose (less current).  The smaller the size of the
capacitor, the smaller the size of your water bucket.

For any given size of resistor and capacitor (hose and bucket), there is a rate
of flow in which you will see the level of the cap (or bucket) rise and fall.
For a smaller resistor (lots of currents) and a small capacitor, it doesn't take
long to see the rise and fall and you can have a higher rate of turning the
signal on and off.  At some point, the capacitor doesn't have enough time to
fully charge and discharge and you don't see the level change as much (filtering
is now in progress).

Increasing the resistor (less flow / current) means it takes longer to charge
the cap (or fill the bucket) and you cannot have as fast of a rate of change (or
frequency).  Likewise, changing the value of the cap will have simular effects
(less capacitance means you can charge/discharge faster and have a higher
frequency).

michael brown wrote:

{Quote hidden}

> This may sound like a dumb question, but I have not run across a good "plain
> English" description of low-pass.  I completely understand low-pass
> filtering with regards to RF and AF.  But when it comes to this stuff, I am
> unsure of what it means.  I am under the impression that it is logically
> related to the process of low-pass filtering AF and RF but not necessarily
> physically related.  By my thinking, it is a way of obtaining a general (but
> accurate) "reading" of some type of signal, without processing all the
> minutia.  If someone could kindly point me to some "understandable"
> description of the process along with some simple examples I would be much
> appreciative.  I am thinking that I have probably incorporated this process
> without "knowing" it.  To me, this is like PID a fairly simple idea with a
> complex definition.
>
> Michael Brown
> Instant Net Solutions
> http://www.KillerPCs.net
>
> "In the land of the blind, he who has one eye is king"
>
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2002\03\01@063852 by michael brown

> Inside a computer a simple low pass filter working on successive samples
is:
>
> Filt <-- Filt + (New - Filt)*FFrac
>
> Where FILT is the filter that is being updated, and NEW is the new sample
> being accumulated into the filter.  FFRAC is the filter fraction, a value
> from 0 to 1 for a normal low pass filter.  It governs how "heavy" the
> filtering is.  A value of 0 is infinitely heavy, which means the filter
> output never changes at all.  A value of 1 causes the input signal to be
> passed thru without any filtering.  Do an example on paper to see how this
> works.  I often cascade two of these things, called "poles", to make a
"two
> pole" digital filter.

Woohoo, way to go Olin.  This is the best description yet, and it even
includes a formula.  ;-D

michael brown

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2002\03\01@084114 by Olin Lathrop

> > >Also I hear/see the term
> > >"integrate" and am not completely clear as to what this means in the
> > >context
> > >of EE and data samples.
> >
> > "Integrate" in the case of discrete samples just means "add 'em up".
> > Same as the averaging technique I mentioned above.
>
> That's simple enough for my pea-brain.  ;-) Why don't they just say
that???

Because there's a lot more to it than that in many cases.  You can only wing
it so far.  You need to go learn some math if you really want to use this
stuff for more than a one-off hobby project.


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Olin Lathrop, embedded systems consultant in Littleton Massachusetts
(978) 742-9014, TakeThisOuTolinKILLspam@spam@embedinc.com, http://www.embedinc.com

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2002\03\01@084133 by Olin Lathrop

> > A low-pass filter is simply a moving-average in a sampled data system.
>
> Ding! Ding! Ding!  The perfect "plain english" description.  I get it now.

Not so fast.  That's only one way to achieve a low pass filter.  In
techno-speak that's called a box filter, which is usually not what you want
to implement in a PIC.  For most ordinary situations, there are better
methods that are actually easier to implement.


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Olin Lathrop, embedded systems consultant in Littleton Massachusetts
(978) 742-9014, KILLspamolinTakeThisOuTembedinc.com, http://www.embedinc.com

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