Low frequency signal generator

That's well within the range practical for direct synthesis with a microcontroller. Well, depending on how good a sine wave you need.

Best regards, Spehro Pefhany

Analog Devices serial DDS chip, maybe, if you don't mind some programming to control it.

John

You might try a four pole switched capacitor filter chip like the National MF-10. You put in a square wave of the fundamental frequency and a sample clock of say 64 time higher frequency. You filter out all (most) of the the harmonics and get out a sine wave .

Dick Hagerman

You might try a four pole switched capacitor filter chip like the National MF-10. You put in a square wave of the fundamental frequency and a sample clock of say 64 time higher frequency. You filter out all (most) of the the harmonics and get out a sine wave .

Dick Hagerman

Dirk Bruere at Neopax wrote in news: snipped-for-privacy@individual.net:

This is very easy with our dspstak boards. You could also do this with our DSP-8300 modules. We have an article that discusses the method and also a working set of examples.

The modules are $125 in 100 qty. A dev kit is available for $250.

How's this for a super chip that does what you describe?

Hi

Looking for a low frequency signal generator chip that can put out a sine wave varying in frequency from around 1Hz to 100Hz with resolution around 0.01Hz Any ideas for a cheap solution?

Thanks. Since it will be feeding an inductive load it doesn't have to be too good. However, I was hoping for some super chip with some switches as inputs, a sine o/p and which drives a 4 digit LCD...

The OP wants "resolution" of 0.01Hz, which for 1 Hz to 100 Hz would require a 10-turn pot (and a VERY accurate dial), voltage regulation and probably temperature compensation as well. Actually I'm not sure what the OP wants, but if the stated resolution is an indication of the accuracy he wants, it surely will take a digital (microcontroller) solution most likely involving an implementation of the "phase accumulator" concept.

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If you don;t need an exact frequency on the output, you can build an oscillator with the switched cap filter acting as the frequency determining components. This removes the need for the extra dividers at the cost of an added dual diode and a couple of resistors.

Wire up both sections as band pass filters

Connect section #2's input to the BP output of section #1.

Take the HP output of section #2, run it through a simple diode limiter and use it as the input to the section #1 filter.

Thanks for all the suggestions guys, and I'll keep them in mind. However, I have decided the best (ie cheapest and hassle free)method is to use a program like Audacity on a PC to generate the signal I need to high precision. I can then either output it directly from the soundcard or convert it to MP3 and play it from a portable device.

I was just checking out with my old copy of CEP what happens when you add a

Well, as far as I know Audacity can create a tone at a very precise frequency. However, you are correct about playing very LF stuff. One solution is to create a stereo track with one channel at f1 and the other at f2 where f1>f2, feed these into a summing op-amp and follow by some simple filtering to get f1-f2.

To solve the LF cutof problem with a normal soundcard, think FM. Modulate an audio carrier say at 10KC/S with the LF frequency required. Stuff the output into a 4046 PLL as an FM detector

martin

Unless you take some steps to calibrate to an external reference, the generated tone can be no more accurate than the quartz crystal timebase in your soundcard. And I have found these crystal oscillators to be off by as much as 1%. That's not very precise. To be fair, most soundcards are better than that. About .02%.

-Robert Scott Ypsilanti, Michigan

That will not work. In order to get (cerate, or generate) a sum or difference of frequencies, one needs to use those signals to drive a nonlinear device. Harmonics are also generated.

Some audio editing software allows you to generate a modulated signal.

Best regards, Spehro Pefhany

No. You have to modulate one with the other, or multiply them with an analog multiplier. In fact, I've heard of an analog multiplier that's optimized for balanced modulator use.

I wonder if that's really the case, if your analog multiplier is "linear". It should be simple for someone who paid attention in that class to answer this one with arithmetic. :-)

True.

Cheers! Rich

Summing them won't work?

The f1+f2 can be filtered out. It should be easy if f1,f2 >> f1-f2