Questions about balanced input and output stages

Hi all!

I'm working on a "stomp box" for microphone, not guitar, so the input is a balanced XLR (dynamic mic) and the output should be suitable for a mixer/PA (balanced output).

The circuit is based around a PT2399, so I need to take out a single-ended signal using an INA333. My design relies on op amps (TLV2462) to amplify the signal etc. All these ICs run on 5 V single power supply, so I need to add a 2.5 V DC offset to signal path.

I am currently focusing on the input and output stages, how to hook up to XLR contacts and feed into the INA333. Similarily, on the output, how to connect the balanced signal (which I create using op amps). Please review my design as shown in the images.

Are the coupling caps on the input stage oriented the right way, with cathode towards the mic? Also, I believe I should have a 48K to ground on the output (just before the XLR contacts) but I should not have this ground on the input stage. The 48K to ground on the output is to stabilize the potential at the output, right? On the input, the potential will be stabilized by the "bias" resistors which add the 2.5 V offset to the signal, so I don't need the ground connection (also, it would create an unintended voltage divider) so these should be omitted.

Input stage
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Output stage
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I guess I also wonder if it makes any difference if the op amps on the output (which create the balanced signal) are in series, or in parallel as in my design?
 
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C2 and C6 in the output circuit should be reversed as their left sides are at the positive bias (artificial GND) voltage.

And in the mic input the coupling caps should have GND referencing resistors (100k) on their left sides so they can charge no matter if something is connected to the input and whether it is ground referenced or not (dynamic mics usually have floating voice coils). Basically the same idea as it it used for the output.
 
Is there any risk that either side will ever be accidentally connected to something with 48 V +/- 4 V phantom power? If so, 63 V bipolar electrolytics may be a better idea.

INA333 has a rather high voltage noise when you want to use it as a dynamic microphone preamplifier: 50 nV/√Hz where the thermal noise of the microphone itself is only 2 nV/√Hz if it has a resistance of about 200 ohm.
 
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And in the mic input the coupling caps should have GND referencing resistors (100k) on their left sides so they can charge no matter if something is connected to the input and whether it is ground referenced or not (dynamic mics usually have floating voice coils). Basically the same idea as it it used for the output.
So I should basically move the 48K to ground to the other side of the coupling caps? Thanks for replying!
 
Is there any risk that either side will ever be accidentally connected to something with 48 V +/- 4 V phantom power? If so, 63 V bipolar electrolytics may be a better idea.
But should the caps on the output be oriented the other way around, given the fact that the mixer might have phantom power? If I design for the case when there is phantom power, will the same design work just as well in the case when I use it on a mixer which doesn't have phantom power?
INA333 has a rather high voltage noise when you want to use it as a dynamic microphone preamplifier: 50 nV/√Hz where the thermal noise of the microphone itself is only 2 nV/√Hz if it has a resistance of about 200 ohm.
Another thing I forgot to mention, the mics I'm designing for are dynamic (not condenser, so no power in mic) and 600 ohm. Are you saying that the INA333 will cause audible noise? Because I am hearing an annoying hum in the system, which is sort of the stuff I'm focusing on now.

My current prototype is working, I can sing through it and it produces nice echo 😊 But there sound quality can be improved. However! If there is some degradation at the end, I can accept that and it will just be part of the "style" of my custom echo fx 😉 That said, I am doing this to learn about sound and analog electronics, so I want to fix at least the obvious errors.
 
But should the caps on the output be oriented the other way around, given the fact that the mixer might have phantom power?

When you use bipolar electrolytic capacitors, like I suggested in post #3, there is no polarity, so it doesn't matter.

The INA333 will cause hiss, not hum. Hum can be due to a poor power supply, ground loops (although balanced circuits are fairly insensitive to those when you make them according to the AES48 standard), poor layout and lack of shielding - all things we cannot see in the schematic.
 
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Right, yes. It's not easy to categorize the background noise, but there is definitely something other than the normal silence one gets when plugging in a mic directly. Maybe this is too much to expect, that any active circuit would be "silent", comparable to just a passive mic? Anyway, lack of shielding is very likely a factor! My current prototype is spread across three pcbs, which are interconnected with JST sockets (cables) and just sitting open on my desk. I plan on tidying this up as soon as I get these basic fixes done.

I am planning on getting film/bipolar caps (with higher voltage rating). However, 10uF is just a random value I chose... Perhaps someone has a good recommendation on a proper value?

Regarding all this, I'd really like to find some learning material covering input and output stages specifically. I have searched, but can't seem to find anything which really goes into this (like, series resistors and impedance matching, coupling caps, where to have resistors to ground, etc). I've looked at the Small Signals book quite a few times but is a bit overwhelming ;-)
 
Doesn't this look like a decent component? It's small/compact, high DC voltage rating, and not expensive 🙂

Film Capacitors 100VDC 10uF 10% CUT LD
Panasonic
https://no.mouser.com/ProductDetail/Panasonic/ECQ-E1106KFZ?qs=KRpiEsko4Hmyt%2BrPXyLDiw==

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Or these, from WIMA:
Film Capacitors 10 uF 63 VDC 10%
https://no.mouser.com/ProductDetail/WIMA/MKS4C051006B00KSSD?qs=K1lkoHpmGTqkuKPa5x9hpg==
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I have some of these lying around from way back -- never knew these square, plastic capacitors were different than other unpolarized caps, like ceramic ones... So these from WIMA are typically (always?) film/bipolar?
 
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Regarding all this, I'd really like to find some learning material covering input and output stages specifically. I have searched, but can't seem to find anything which really goes into this (like, series resistors and impedance matching, coupling caps, where to have resistors to ground, etc). I've looked at the Small Signals book quite a few times but is a bit overwhelming ;-)

For mic or line input you are looking for a 'bridging' topology.
It means an input impedance many times higher than the output impedance of preceding stage ( you transmit a voltage rather than a current through the transmission line).
For a mic input typicaly the range is 4/5 times the mic output inpedance so circa 1200/3000 r input impedance. As there is different kind of microphone technology this ratio can sometimes vary ( eg: with a ribbon mic ratio is way higher) but still principle remain.

For line you can see up to 10 times or more ( circa 10k input z or more).

About coupling caps the bandwidth needs to go down circa 20hz. Just use basic calculation of filter to determine the value.
That being said, 20hz bandwidth on a mic input can be bothersome as mechanical noise ( handling noise) will happily be capted by capsule so it might be wise to introduce hpf choice. Freq depend on application and can sometimes help to tame proximity effect with directional capsule ( hp filter on some mic is located to counteract this from time to time).

Take a look at Jensen's transformers whitepaper as well as THAT mic preamp IC white papers to have more in depth explanations about mic situation. Look for the ones where Bill Whitlock is authors both on Jensen and THAT sites.
Look for the one about THAT 1606/1646 series too whith same author about line.
 
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Thanks for responding, and for the helpful information. Of course, the capacitance forms a filter which gives me a metric to choose a capacitor value. So much to keep in mind, I had forgotten this 🙂

I'm using INA333 for differential input, which also serves as a preamp: RG of 1K gives a gain of ~50, which (according to my "estimates") should amplify the signal from the mic to something in the working range of the PT2399. Following the PT2399, the "wet" signal is mixed with the dry signal (using a 20K pot) before everything is attenuated by 1/50, to bring back down to "mic level" (i.e. undo the initial amplification). All op amps used are TLV2462. In the output stage, I also use TLV2462 for a makeshift line driver, to produce a (hopefully) balanced signal to the mixer.

I would post the schematic of my FX stage, but I'm afraid to go too far outside the original scope of this thread (the input and output stages, specifically)...?
 
Could you give a rough explanation of what you are trying to do, it could help to understand your plan. Schematics too. 😉

As far as i understand you want to make a preamp to enter the pt2399 and then use mic input on your desk?
If so why going back to mic level ( and go though another mic preamp) rather than using line input on your desk?
 
Hopefully this shows the general idea of what I'm trying to do (and hopefully it doesn't contain too many embarrassments for me). Keep in mind, this is a hobby project 😀

This is to be a "stomp box" for a mic (similar to guitar), so the user should be able to toggle effect using S1. The bypass is handled by a digital switch (U4). Signal path (when S1 is on) is through INA333, which also amplifies x50. Then, a 2.5V DC offset is added to the signal, since all ICs run on single power supply (5V). The buffered signal is sent to the FX subcircuit (see separate image), and the return (wet signal) has DC offset "re-applied" (I could probably remove the caps on the PT2399 output, but I've kept to the original schematic from the PT datasheet). A 20K pot handles dry/wet mix (blend), before the signal is attenuated back to mic level, and sent through a DIY line driver (using two op amps).

The reason I'm attenuating the signal is because this is a stomp box effects circuit. If I didn't attenuate, what would then happen when the user toggles off the FX, and we'd effectively send mic level directly to the line input? That would probably result in very low sound, or am I totally off here?

I'm wondering if I maybe don't "need" the U7A buffer, following the INA333...?

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