Subaudio Multiplier for Soundcard Input

Are you saying there is Windows software available that will demodulate an AM signal fed into the soundcard on the fly?

Mark Harris

Reply to
mharris
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It's worth a try then. Cheaper, for some reason, than a USB DAQ.

Maybe someone could make a comfortable lving converting these to resell at half the price of a "real" DAQ.

Mark Harris.

Reply to
mharris

Do you mean the citrcuit I have designed will do this "as is". If not, what would you suggest changing?

BTW I had previosuly considerd using an MLT04.

Do I understand correctly that what you described in the first entry is not considered to be a "frequency multiplier? What would you call it?

Mark Harris

Reply to
mharris

How would the acquired FM signal appear in the spectrograph software as a readable relpcate of the orignal ELF?

Mark Harris

Reply to
mharris

Of course. Look at SDR (sound defined Radio)..

But in any case, you can have the sound card monitor a steady tone and that tone can be a AM carrier.

Multiple reads of AM levels of a constant tone on the input can be used to regenerate a stream of PCM values via the amplitude of the tone. A dead carrier would be a tone sitting in a steady state around 50% of the full amplitude. doing a +/- amplitude can allow you to regenerate the original ELF signal, right down to 0 HZ.

But I made a comment earlier about some sound cards. AGC's can mess with you. It works best if you could generate a marker/sync pulse at full amplitude and then generate the actual signal at lets say 75% for

100% modulation. This way decide in software which is which.. But if you think you can insure the softcards settings to not use AGC then you should be able to set the input levels properly. Jamie
Reply to
Maynard A. Philbrook Jr.

It's a voltage multiplier.. one input multiplies the other input and the sum is at the output.

However, don't get to upset about this, it will generate DSBs, fundamentals, third harmonics from the square wave etc..

it'll end up sounding like Jimmy Hendrix..

Jamie..

Reply to
Maynard A. Philbrook Jr.

Do you mean "Software Defined Radio"?

Reply to
krw

Because of the cap, I specifically mentioned this usb soundcard where the PCB is easily exposed. And I also mentioned a level shift would be required.

Reply to
miso

A full amplitude carrier AM modulation demodulation can be done by simply

1a.) take the absolute value of each sample (full wave rectification) 1b.) set any negative sample value to zero (half wave rectification) 2.) low pass (maybe 150 Hz) signal to remove carrier residuals 3.) high pass signal (say at 0.1 Hz) to remove DC bias

Phase locking to the reduced carrier can be hard when the audio sidebands are only a few Hz away from the carrier.

Maybe a Costas loop might function for a DSB signal.

A full amplitude a few samples long synch pulse every 50 ms should do the trick. Should be easy to detect and eliminate from the data stream and compensate against AGC gain changes. Since the antialias filter will broaden the pulses, a bit longer section needs to be cut out from data stream.

It might be a good idea to invert every other pulse to remove any DC bias.

Reply to
upsidedown

It generates double sideband suppressed carrier signal.

To generate normal amplitude modulation (A3E), add a DC bias to the audio input terminal on the multiplier, so that the voltage is always positive compared to the reference input. This will enable the carrier to pass through all the time and hence taking the absolute value of the modulated signal can be used for detection.

The analog input should be low pass filtered well below the carrier frequency (actually below the lower sideband). A sine (not square) wave should be used as a carrier, to avoid sidebands and carrier around three times the carrier frequency.

It is a _voltage_ multiplier.

Reply to
upsidedown

Exactly as the AM sidebands provided that the modulation index is kept extremely low, to prevent the creation of second order sidebands, which would overlap the first order sidebands.

Reply to
upsidedown

The OP did not say that the ELF signal is digital.

DSB is difficult, as the re-inserted carrier must have both proper frequency and phase. For a signal without a recognizable sync sequence, the loop is prone to mis-lock.

--

Tauno Voipio
Reply to
Tauno Voipio

Since U2 seems to have differential inputs for both X and Y, applying a DC bias on X2 should help keeping the X2 (audio) to X1 (bias) difference always at positive or always negative level and hence generate normal AM (A3E).

Reply to
upsidedown

Yes, excuse me..

Jamie

Reply to
Maynard A. Philbrook Jr.

Big snip Reticon - 1024SAD ???? - it's an analog delay "bucket brigade" chip. Company is no longer around but chips of this type were/are really common as a sound effect for guitars. All digital versions are very common in the music industry. For example there are boxes that can be set to generate any harmony (x2, x3 whatever), for a vocalist, with excellent fidelity. You can find the same function built into alot of audio recording software which allows you to change tempo or key or both. bg

Reply to
bg

Digitally you can do just about anything, it's only a matter of horsepower and available budget.

More esoteric analog tricks would be, for example, deliberatly distorting, picking one of the harmonic sets and downconverting that. Now you'd have a 2x, 3x, 4x ... 10x or whatever stretch. I just don't see how that would make sense in Mark's case.

--
Regards, Joerg 

http://www.analogconsultants.com/
Reply to
Joerg

On Sat, 07 Dec 2013 09:13:39 +1100, snipped-for-privacy@comprodex.com wrote:

--- You can do it with the analog multiplier you have on hand, but just for fun:

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Reply to
John Fields

That would work if the ADC633 was happy with some common-mode voltage. Some chips with differential inputs really want them to be _differential_, without a lot of common-mode voltage on them.

I have no clue if the ADC633 is such -- that's what data sheets are for.

--
Tim Wescott 
Control system and signal processing consulting 
www.wescottdesign.com
Reply to
Tim Wescott

Thanks, John

I will have a close look at this after work.

Mark Harris

Reply to
glewis

OK. I will spend some practical learning time at the workbench on this.

Lots of options to choose from, thanks to the generous responses so far.

Mark Harris

Reply to
mharris

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