Passband Puns

I'm using MATLAB's RF Mixer:

RF Mixer with added out.simout

The RF Mixer creates a signal and my overall aim is to STFT this. Usually the outport is set to Complex Baseband or In-Phase and Quadrature values as the output, which from the scope looks like this:

https://preview.redd.it/l2a6m4i8gzp61.png?width=548&format=png&auto=webp&s=f1c33dab80baba98f07210de9246fdd2003b65cf

However, the STFT function needs an input vector describing that signal and since it's in the time-domain only real values should be used (I think this is right), so I used Real Passband values in the RF Outport, with the aim that at each step, this value should be concatenated to a vector and at the end of all steps, I can STFT that vector. However, the Real Passband value looks like this:

https://preview.redd.it/laqi4a7ogzp61.png?width=563&format=png&auto=webp&s=ff261da8ab73976fadbefebb97365658d7d1abf0

This looks wrong to me since I shouldn't need the imaginary part to describe the waveform? And if it is wrong how can I ensure that my STFT will be using the correct values?

A friend who has an IC-7610 found that when he tuned to an FT8 fox-and-hound band he was unable to hear the DX, who was operating down around 300 Hz on the waterfall. If he then manually set the 7610's tuning about 500 Hz low, he could now see the DX. The 7610 was in usb-d mode.

I'm not familiar with the 7610 and its filters, but I used to use a 7300 and never ran into this issue. If the radio is set to usb-d, is it still possible to have a filter selected that blocks the few hundred Hz closest to the tuned frequency? What's the best setting for the 7610 to resolve this?

Hello AskEngineers

My book has a proof for the Nyquist rate of a baseband signal and it proves that you must sample at at least twice the maximal frequency. The proof strategy is using fourier transforms to show that you can use a filter to get back to the original signal if you sample at least at this rate. I think this is probably the proof that everyone has seen when they were introduced to the Nyquist limit.

Now, my book then mentions (without proof) that when we will talk about passband signals in later chapters that we can expand the theorem to using at least twice the bandwidth B instead of twice the maximum frequency.

In an exercise the TA and a few fellow students then applied this to an audio signal (in baseband) where the minimal frequency is 1 kHz and the maximal frequency is 3 kHz. They used 2B as the minimal sample rate. When I try to apply the fourier transformation logic used in the proof it shows that you won’t actually be able to recover the original signal that way. (see this drawing)

I suspect that you can only use 2B if the minimal frequency is significantly higher (probably at least B?) than the bandwidth. Basically if you actually have a real passband signal and not if it’s just a baseband signal with a few lower frequencies missing. Is my suspicion correct?

Thanks for reading and for any input you may provide.

It is for use in a superheterodyne NMR spectrometer. I have tapped out the usefulness of pre-mixdown RF amplifiers. I found I can benefit from a high impedance low freq. amplifier to be placed after mixdown, between the receiver and the scope - here the desired signal is down to about 10 KHz give or take.

I can piggy back two amplifiers with a gain of around 10. I'd enjoy being able to sweep the gain or at select from some values.

I am looking at this pretty standard non inverting gain amplifier. But I need band selectivity and I gotta block DC. I also need a high input impedance.

1) How do I achieve the freq. band desired?

2) Where do I put blocking caps? How big?

3) This bitch has got to be clean. Is g=10 too greedy? I'd like to get up to 100. Can I put them in series without running into problems?

4) Gain sweep/selector? Can I just use a pot or should I sweep the DC input?

5) Which op-amp? Model No?

I have virtually limitless resources. Just have to attempt at low cost and of course minimize complexity for the sake of, well, obvious.

Thanks reddit!!

Say I use a SQM to measure the colour of the sky using three colour passband filters, using that aquired data, can I then map the colour of the sky onto a CIE colour map? (ie the 1931 colour map).

Sorry if this is the wrong place for this.

All documentation I can find requires the filter be designed as a filter type object so that the above parameters can be measured. Must the filter be in that format to get various parameters quickly? If so is there a conversion function to define your designed filter as an object? Otherwise, must all the parameters be calculated manually? Also, aside from freqz and impz, what other useful functions are there for filter analysis? Thanks!

I've been looking at this circuit and I understand that the passband gain is A = -R1/R2.

However, from the transfer function of this circuit:

sC2R1/((1+sC2R2)(1+sC1R1))

I understand how the 3dB breakpoints are calculated, but I'm not sure how the passband gain was derived. Could someone point me into the right direction?

Go post NSFW jokes somewhere else. If I can't tell my kids this joke, then it is not a DAD JOKE.

If you feel it's appropriate to share NSFW jokes with your kids, that's on you. But a real, true dad joke should work for anyone's kid.

Mods... If you exist... Please, stop this madness. Rule #6 should simply not allow NSFW or (wtf) NSFL tags. Also, remember that MINORS browse this subreddit too? Why put that in rule #6, then allow NSFW???

Please consider changing rule #6. I love this sub, but the recent influx of NSFW tagged posts that get all the upvotes, just seem wrong when there are good solid DAD jokes being overlooked because of them.

Thank you,

A Dad.

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