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Firstly, if this is the wrong place for this, I'd be happy to know the correct place for this and I'll move it there.

I have two PCs which share an audio interface. My laptop is connected via USB and uses it as its sound card. My Desktop (runs Linux so will be called Linux PC from here on) inputs into inputs 3 and 4 of the interface which is routed out the monitoring outputs. If it's relevant, I'm using a Focusrite Scarlett 6i6 (2nd Gen). I was watching something on YouTube on the Linux PC about an experiment involving sound, one of those things that let you try at home. Interested, I opened my DAW on my laptop, set it to read inputs 3 and 4 (the Linux PC), and opened a spectrum analyser to take a look at the frequencies of sound.

My spectrum analyser shows the audio has a pulsing every 10 seconds with the pulse lasting around 7 seconds, with two pulses in unison at ~20kHz reaching up to ~24kHz and the second at ~40kHz reaching up above the scale which tops off at ~43kHz. My sample rate is 88.2kHz.

I've included a screenshot. The left is a spectrogram before applying a High Pass filter (via EQ) set at around 18kHz, the right is after filtering. enter image description here

What is this?

I've looked at this question and accepted answer, which is the only thing I could find that has come close to be even remotely similar to what I'm seeing and I don't quite know if it's related or not. The post mentions to filter out anything above 1/2 the sample rate to filter out aliasing artefacts, but 1/2 my sample rate is 44.1kHz which plants the lower pulse well below this area and the higher one just below it.

The screenshot was taken when (seemingly) no audio was playing. I can't tell if the plugin has some kind of automatic gain control that makes the maximum amplitude fit in the spectrum's colour gradient or not, but when audio plays it is significantly weaker on the spectrum but definitely still there.

I've also recorded, intensely amplified and lowered the pitch and it sounds like data, but again, what data?

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  • $\begingroup$ Have you measured your laptop audio, 6i6 (use RMAA software) and PC audio (is it integrated audio chip in question) spectrum's separately? $\endgroup$ – Juha P Jan 17 at 8:21
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    $\begingroup$ Most likely your sound card is bandpass sampling EMI generated by a smart utility meter installed by your utility company. The duty cycle and period remind me of a Neptune water meter $\endgroup$ – Andy Walls Jan 17 at 14:06
  • $\begingroup$ Would you make the recording available please? $\endgroup$ – Olli Niemitalo Jan 17 at 15:44
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Not really an answer but some hints on how to proceed.

  1. You need to debug this on step at a time. Start with shortening the inputs to the Focusrite and disconnect the Linux computer entirely. If you still see noise, disconnect the power supply from you laptop and run battery only. If it's still noisy, either the Focusrite or the USB interface are bad (i.e. ground provided by your USB port is just too dirty).
  2. Then connect the Linux PC ground only (leave the inputs shorted).
  3. Then connect signal as well but make sure the output of the Linux PC is all the way down and verify by measuring: should be zero volts.
  4. Then play a -20dBdFS sine wave at 1kHz from the Linux PC. Compare that to expected results.

The fix will depend on where in the chain things go wrong. Audio connections are tricky and susceptible to power supply contamination, ground loops, Electro Magnetic Interference, etc.

Hence you should used balanced connections whenever possible. Try using the XLR inputs on the Focusrite by either using a properly configured "mono to balanced" adaptor cable. Means that you should NOT connect signal ground. You can play around with connecting chassis ground or not. The easiest solution would probably be to use a passive audio isolation transformer with a ground lift switch. Something like this perhaps: https://www.audiopile.net/FDB-202 (I'm not affiliated).

Good reads about this: https://www.ranecommercial.com/kb_article.php?article=2107

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