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Precision low frequency generation

  • Thread starter Dirk Bruere at NeoPax
  • Start date
M

Martin Brown

So what would I get from linearly adding two different frequency sine
waves?
When I do it in Audacity (s/w) I get a signal modulated at the
difference frequency.

The mathematical identity :

sin(A) + sin(B) = 2 sin((A+B)/2)cos((A-B)/2)

Comes in useful for working this out.
Substitute A = wt, B=ut to see how the frequencies produced by linear
mixing are related.

w, u added together in equal amounts makes a (w+u)/2 carrier amplitude
modulated (multiplied) by (w-u)/2.

If you wanted to generate a pure sine wave at a low frequency you
multiply two similar frequency sine waves together. However, I reckon
a sample and hold would be neater and easier.

Regards,
Martin Brown
 
P

Phil Allison

"Martin Brown"
The mathematical identity :

sin(A) + sin(B) = 2 sin((A+B)/2)cos((A-B)/2)

Comes in useful for working this out.
Substitute A = wt, B=ut to see how the frequencies produced by linear
mixing are related.


** There are absolutely ** NO ** new frequencies produced by your so called
"linear mixing" - correctly described as " summing " .

The identity quoted above does NOT describe amplitude modulation.

It describes a quite different phenomenon called "beats."

Look it up.

You bloody need to.



....... Phil
 
J

John Larkin

I'm a newbie-lurker to this group, and enjoy the subject
matter because people are usually polite or flame with
a joke, but I think Dirk have pressed the boundaries of
etiquette.
Oh, and BTW, I'd use a phase-shift oscillator.
Ken

Phase-shift oscillators are textbook darlings that usually work badly
in real life. Lots of drift, lots of distortion, hard to tune.

John
 
K

Ken S. Tucker

On Nov 21, 1:39 pm, John Larkin > >
Phase-shift oscillators are textbook darlings that usually work badly
in real life.

We're doing 2.5 Hz oscillator right?
Lots of drift,

Component quality, thermodynamic stability,
typical of oscillators.
lots of distortion,

That depends on the amplitude.
With anything, keep the amplitude low and
within the linear region of the feedback transistor
and you can get into spec.
hard to tune.

Not really once you're in the ballpark, a pot
tweaking the top of one of the resistors will
do ya.

If I was told to make it ultra accurate I'd phase
lock loop off a crystal in a thermally controlled
oven. That way getting 1/10^5 is fairly easy and
1/10^6 is manageable. I find 1/10^8 a bit harder,
but that's a drift of 1 hz at 100 Mhz.

While with Telefunken, we had some 100Mhz
oscillators sent from Germany, (fancy units,
based in crystal controlled ovens), and they
were off 2 or 3 hz, and the damn problem landed
on my desk. From relativity to altitude pressure,
I couldn't figure it out, so I had them re-adjusted
(tweaked), following SOP I blamed Germany.
Of course that's nothing compared to the fella's
who have a dozen atomic clocks that need to
be synchronized.
Fun Stuff
Ken
 
M

Martin Brown

"Martin Brown"





** There are absolutely ** NO ** new frequencies produced by your so called
"linear mixing" - correctly described as " summing " .

But there is an equivalent description of it as an amplitude modulated
mid frequency. And more imporantly here if you synthesise the right
hand side of the equation using a multiplier you can get what you want
on the left hand side by filtering out the unwanted high frequency.
The identity quoted above does NOT describe amplitude modulation.

I note that you have deliverately snipped the context.

It does for the case where the modulation is by a lower frequency sine
wave on the carrier.

A(t) = m(t)sin(wt)

Normal AM modulation would insist that m(t) > 0 and relatively slowly
varying to keep the sidebands controlled.

And it suggests that a fairly cute way of getting what the OP wanted
would be to tweak a classic FET square law AM modulator design to
match his requirements. At these low frequencies it shouldn't be too
hard. eg

http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/RadCom/part9/page1.html
It describes a quite different phenomenon called "beats."

Perhaps when you get out of secondary school you will learn how these
things are related.
Until then GROW UP!

Can someone please explain why Dirk Bruere has a net stalker called
Phil?

Regards,
Martin Brown
 
P

Phil Allison

"Martin Brown = Fucking MORON "

But there is an equivalent description of it as an amplitude modulated
mid frequency.


** Like *hell* it is " amplitude modulation ".


I note that you have deliverately snipped the context.


** I carefully note the YOU are a fucking nut case LIAR !!!!

Perhaps when you get out of secondary school you will learn how these
things are related.


** YOU really need to get your pointed, ASD fucked head out of you fat
stinking ARSE for a moment - just to see how fucking WRONG you are.


Until then GROW UP!


** That is an OUTRAGEOUS lie !!

Go get a nice big dose of bowel cancer

- you vile, pile of sub human shit.


Autistic cunts like YOU need to be rounded up and exterminated.

Win first.




....... Phil
 
D

Dirk Bruere at NeoPax

But there is an equivalent description of it as an amplitude modulated
mid frequency. And more imporantly here if you synthesise the right
hand side of the equation using a multiplier you can get what you want
on the left hand side by filtering out the unwanted high frequency.




I note that you have deliverately snipped the context.

It does for the case where the modulation is by a lower frequency sine
wave on the carrier.

A(t) = m(t)sin(wt)

Normal AM modulation would insist that m(t) > 0 and relatively slowly
varying to keep the sidebands controlled.

And it suggests that a fairly cute way of getting what the OP wanted
would be to tweak a classic FET square law AM modulator design to
match his requirements. At these low frequencies it shouldn't be too
hard. eg

http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/RadCom/part9/page1.html




Perhaps when you get out of secondary school you will learn how these
things are related.
Until then GROW UP!

Can someone please explain why Dirk Bruere has a net stalker called
Phil?

I think I dissed his posse once upon a time.
Who knows? [Maybe not even Allison]

Dirk
 
P

Phil Allison

"Dirk Bruere at NeoPax"
So what would I get from linearly adding two different frequency sine
waves?
When I do it in Audacity (s/w) I get a signal modulated at the
difference frequency.


** Wot an utterly, monumental fuckwit.

This vile psychopathic pile of **SHIT** must be exterminated from
the face of the earth.

http://www.neopax.com/

http://www.neopax.com/dirk bruere.jpg

http://www.neopax.com/asatru/index.html


It may take any number of wooden stakes, driven through this vile zombie's
heart.

Show no mercy - it ain't human.



....... Phil
 
R

Robert Baer

You need to define precision.

How about this. You can make a simple sine wave generator feeding a
square wave to a switched capacitor filter. Now if the clock for the
switched capacitor filter was the output of the sound card, you would
get a much lower frequency sine wave than the clock. The square wave
generator is made by dividing down the switched capacitor filter
clock.

Let's do a simple example. The sound card is putting out 100Hz. Let
the clock to corner of the filter be 100. Now you have a filter at
1Hz. Divide the switched capacitor clock by 128 to make things simple.
This gives you a square wave at 0.78125Hz. This is in the passband of
the switched capacitor filter. The first harmonic you need to worry
about is the 3rd, or 2.34375Hz. Use an elliptic switched capacitor
filter with a transition ratio less that 2, and the 3rd harmonic and
the remaining harmonics will be filtered.

The frequency of the sine wave will be very precise. Fidelity is
another story. You would be doing well to get about 70db down.

There are trick you can do to reduce the harmonics fed to the switched
capacitor filter. One simple trick is to feed a signal that is 1 1 1
0 -1 -1 -1 0 , that is, reduce the size of the jump, which in turn
reduces the harmonics. You could also generate a signal with resistor
taps to reduce the jumps further.

If you have a bigger budget, use a digital sine wave generator scheme
(dac and rom), where again the clock is from the sound card.

I saw a DAC on the parallel port mentioned. This doesn't work well
unless you use DOS. The port control timing is not very good in modern
operating systems.
Port timing could be precise *if* the verdammdt timer tick (aka RAM
refresh) was not NMI.
 
R

Robert Baer

Dirk said:
Because ultimately I also need the output to follow complex time
dependent frequency patterns eg 6Hz to 2.5Hz over a ten minute period,
followed by 2.5 to 8 over three minutes, followed by... etc

Dirk
Then use a microcontroller - MILLIWATTS (not 100+ watts).
 
R

Robert Baer

Dirk said:
So what would I get from linearly adding two different frequency sine
waves?
When I do it in Audacity (s/w) I get a signal modulated at the
difference frequency.

Dirk
A *linear* system *CANNOT* create anything not given to it; the
output will be the arithmetic sum or differences of the inputs (only
thing added is the noise of the circuit).
Software digitizes, it IS NOT LINEAR.
 
you are a perturbed person a computer to generate frequency? precision ? give us a break dream of something else that makes sense.
 
J

JosephKK

Dirk Bruere at NeoPax [email protected] posted to
sci.electronics.design:
Thanks.
Initially I need to test with sinewaves, but ultimately I want to
use the computer to generate arbitrary LF waveforms.
Modulating a HF signal with the LF waveform in the computer would
seem the best way to go initially. Then it becomes an AM 'radio'
problem. Cat's whisker?

Dirk

After reading all this is still sounds like you want a ultra long
record 2 to 20 Hz arb generator, based on a micro, with a web
interface. Then the PC can do the heavy lift waveform synthesis,
send the wave to the arb and out it goes. Memory is cheap, a couple
gigs of flash and a hundred megs of ram buffers and a micro or an
FPGA and you are in business. USB and Ethernet interfaces to boot.
 
I'm thinking of creating a precision low frequecy generator (<20HZ) by
taking a computer generated stereo sinewave output with (say) one
channel o/p 5000Hz and the other (say) 5002.5Hz, feeding into a
summing input of an op amp and then putting the signal through a low
pass filter to recover a 2.5Hz sine wave. Do you think this will work,
or have I missed something?

Dirk

Yeah use a subtractive amp.
 
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