Demod PC Generated ELF

I have a simpler solution: remove the DC blocking capacitors from the soundcard and create an output stage that uses the reference (usually half the supply voltage) of the soundchip.

--
Failure does not prove something is impossible, failure simply
indicates you are not using the right tools...
                     "If it doesn\'t fit, use a bigger hammer!"
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What is wrong with a digital PCM output signal.

Bob

If you only need one output, you could use both channels of the sound card and feed a mixer to make the demodulated sub-audio. This could be more linear than just rectifying one channel to make the demodulated signal. I would base the demodulator of the HC4053 analog switch.

If I was taking the signal from the sound card, the first thing I would do would to high pass filter it to remove any low frequency noise that the PC may be putting on it. Then you know that the output would be all the demodulated result and not feed through of the noise.

You can also buy DACs that can be interfaced to a PC via USB or RS-232 This may be a better way to go.

Are you referring to a balanced demodulator as described here?

I gather both an external mixer and demodulator are required. Is this available in one IC?

Could you please describe in a little more detail how the technique you kindly described would work with the software/soundcard approach in my OP?

I did a web search but found mostly RF apps.

I was planning to use 1KHz as the carrier for the ELF. According to your suggestion, what would be the recommended cutoff frequency for the high pass filter?

Robert Davis

"Robert Davis"

** You hope to do this or you ARE doing this now?

** You hope to do this or you ARE doing this now? ** A half wave, precision rectifier is all you need to demod AM.

See: " Improved Circuit ".

..... Phil

Ah, we go down to DC with sound cards all the time.

This model is a not bad, just add a DC offset circuit:

|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A1|293%3A1|294%3A50

Then something like cool edit pro to make LF wav files using the built in math function.

Steve

INA---+---\\/\\/-----+-----/\\/\\------ ! ! ! ! ---- ---!-\\ ! ---!O ! ! >--------+---- !

[..after interruption..]

No, this is just a way of using the two channels of the audio to make a lower frequency. One channel is the reference and one contains the information.

You can see the effect if you can fly a spreadsheeet.

A1 is 0 A2 is =3D0.001 + A1 Copy A2 and paste into the rest of the column

B1 is =3Dsin(A1) Copy and paste to the rest This is your intended signal

C1 is =3Dsin(100*A1) Copy and paste The is the "reference" signal

D1 is =3DC1 * B1 Copy and paste This is the modulated signal

C and D represent what will come out of the audio card. Now we can make what the circuit does.

E1 is =3DC1>0 Copy and paste You get many TRUE FALSE cases. In the circuit they are 5V and zero.

F1 is =3DD1 * 2 * (E1-0.5) copy and paste This relies on TRUE being considered 1.0 and FALSE being considered

0.0 by the spreadsheet.

Now if you graph B and F you will see that F is sort of like someone added a bunch of fast stuff to B. A simple low pass filter and some in will give you back almost exactly B with a little lag in time.

Is this

Monitoring global warming? (duck)

not without bypassing the output capacitors, a fairly easy task.

seems like doing it the hard way.