Building a cable from sound system to "mic in"

[kenaycock]

Any updates on this headset buddy line level to mic level adapter shared earlier in the thread?
https://www.amazon.com/Headset-Buddy-Li ... 00OAW85ZG/

On a whim, I tried plugging line level directly in to the mic input on my lappy. Much less complicated than plugging into the TRRS jack on a phone. Without attenuation, the distortion is... lovely.

This also looks like a possibility:
https://www.amazon.com/Movo-MV-RC300-Mi ... 073HR6SY4/

[russellhltn]

If you're using a laptop, I'd recommend the Sabrent USB External Stereo Sound Adapter for Windows and Mac.. That and a plain old 3.5mm stereo cable will work fine. It avoids a whole bunch of ifs, ands, buts, and maybes.

But before placing the order, dig into the hardware settings in the control panel. You find find a "mic boost" setting. If so, set it to 0dB and try again.

At $8, it's also one of the cheapest options.

[Mikerowaved]

I tried the Headset Buddy cable, but found it had too much attenuation for the phones we were using. I'm working on a custom L-pad that will bring the level approximately up to that of the Tabernacle Choir when they come on, but still have a reasonable source impedance for the phone. I'll let you know what I come up.

[mcallaghan]

This project met a quick end. I went to the meetinghouse last evening and discovered we don’t have a record out plug.

Yeah, that doesn't go well.

The plug can be hidden. I've found them inside the podium, as well as inside a cabinet near the "clerk's desk" on the stand.

I still haven't found one in our building. We don't even seem to have an ALS system that I can tap into.

[Wattsuk]

What model amplifier and preamp/mixer do you have fitted?

[leftofdamascus]

If you're using a laptop, I'd recommend the Sabrent USB External Stereo Sound Adapter for Windows and Mac.. That and a plain old 3.5mm stereo cable will work fine. It avoids a whole bunch of ifs, ands, buts, and maybes.

This didn't work in my meetinghouse. The bishopric manages Zoom Meetings from a laptop on a stool/box at the side of the pulpit, in front of them so they can see the speakers face on screen, via a camera just in front of the podium (on a stick attached to hymn books so it doesn't move when persons at the pulpit shake the tabletop of the podium). The audio signal going from the side of the pulpit into the Sabrent AU-MMSA would intermittently come through and fade out and come back. It was even doing it at the A/V rack at the back of the building (out of an RDL D-series PSP1 speaker, hooked up to the same RDL distribution amplifier as the signal coming to the Sabrent AU-MMSA USB Sound Adapter [audio interface]), but only when I plugged into that Sabrent device.

Sabrent tech support mentioned that their device expects, in the pink mic port, an unbalanced signal, mic level, with the positive signal on the tip at 2kohm [or presumably less] impedance, and the sleeve as ground, and the ring isn't hooked up or maybe shorts to ground, but is used in other devices for stereo signals. The Sabrent device puts 2-5V DC bias voltage on the tip, for electret microphones, and they recommended using a DC blocking capacitor for my situation. It's important to note that DC bias voltage can saturate a transformer, causing it to malfunction, which can even cause issues at the source device feeding the transformer.

EmTech tech support wouldn't comment on whether the MSC-U device (that outputs our audio from the side of the pulpit) uses professional line level, or consumer line level, and they wouldn't comment on whether the ground is just passing through, or whether it goes through its transformer at all. They mentioned that EACH signal wire is 600 ohm (a curious statement that made sense later).

RDL tech support mentioned that our audio distribution amplifier ( RU-ADA4D ) uses differential balanced output (phase inverted signal on negative wire, and positive and negative wires are impedance matched), not just balanced (impedance matched but negative wire doesn't have phase inverted signal). Their literature also mentioned that the device output is isolated (thus presumably it is transformer based output rather than active output). It also mentioned that it can output balanced (really differential balanced; 150 ohm; pro line level) or unbalanced (75 ohm; consumer line level), and its input is to be line level (roughly pro or consumer line level, and a bit beyond those, with an input gain knob on the front, where you dial it in to be green with some flickering red when peaking/limiting).

I couldn't get the input gain to indicate the green and red appropriately, so I eventually had to get the sound engineer who does audio blueprints for buildings to adjust the QSC 110F DSP output to be line level to that RDL D.A. Ideally you email the FM Group about such things rather than file an FIR support ticket that gets routed to Emcor, or similar, who will close out the ticket as completed, because it doesn't apply to them, and they'll charge for it anyway (this has literally happened in my experience).

Once our DSP signal was allegedly at line level, I wanted to find out if the MSC-U transformer had its windings kicking down the levels some amount, so I could know how much to attenuate to get that output to mic level before going into the Sabrent device. This proved problematic because you need sound going from the pulpit mic, or it's wire, to measure the MSC-U output, and how loud should that sound be?

I tried sending a -40dbu level signal (loudest mic level signal; -60 dbu is the quietest that you can have the peak of a waveform in the A/C audio signal; I was using a multimeter, so using RMS rather than peak electrical measurements), but the multimeter was bouncing all over. That's because it was a composite audio signal with many wavelengths overlapping, so then I tried sending a lower frequency sine wave (40-125hz or so, so it hurts my ears less) and that got the multimeter to stop bouncing values. I was sending it from my cell phone headphone trs jack into the crab box (EmTech EJ-8) which has 100 ohm XLR output, using this website ( https://onlinetonegenerator.com/ ), with my phone and that website having their volumes maxed, and turning the crab box volume to dial in the -40dbu, based on ohms and voltage, because I needed the software volume dials to be easily repeatable since websites refresh and lose their volume and ohm values, etc. I used this website for the dbu calculations into voltage, relative to ohms impedance of each device: ( https://www.analog.com/en/resources/int ... nvert.html ). Once the crab box had -40dbu output, from XLR, between pins 2 (positive) and 3 (negative), I plugged the crab box XLR into the pulpit mic's XLR input wallplate (with the pulpit mic XLR removed therefrom), and I turned on the QSC touchscreen at the Bishop's pedestal, and turned up its pulpit mic volume slider all the way (so it's easily repeatable), and checked the levels coming onto the harness that goes into the RDL D.A. input, from the QSC 110F DSP, using the multimeter probes on the harness screws for positive and negative, to see what levels the QSC 110F DSP was outputting, and using the aforementioned website to convert the voltage to dbu relative to the QSC 110F DSP impedance (220 ohms for two wire balanced). I then plugged that harness back into the RDL D.A. input. I then repeated my measuring steps on an empty harness (Euro block/Phoenix connector) that I moved into place where the RDL D.A. output goes to the MSC-U, checking the voltage with probes on its positive and negative screws, and turning the RDL input gain knob until it came out at pro line level voltage (at 150 ohms), with the RDL output dial set to the pro line level (+4dbu) marking on the front of that device. I then plugged the actual wired harness back into the appropriate RDL output (moving the empty harness Euro block/Phoenix connector back to where it had been). Then I plugged in a TRS cable into the MSC-U output to see what levels were coming out of it, to see if its transformer windings kicked down the level, etc. I couldn't get a signal between the tip (positive) and ring (negative) but I could get a signal between tip and sleeve, and also between ring and sleeve. Therefore, it seems that the MSC-U is a balanced line level transformer only on its input: it seems to convert a balanced signal (of whatever impedance and whatever line level) to two unbalanced signals, each of which is 600 ohms. So the ground presumably isn't just passing through from the back to the front of the MSC-U wallplate, but seemingly goes through the transformer. You might regard that as a floating ground in a sense, but not a dead end, as it's part of the output circuit. Because the RDL is presumably transformer based output (it's isolated), you wouldn't hurt anything at the RDL D.A. by shorting the MSC-U output ring to ground, if you presumed it was balanced output and wanted to try to convert it to unbalanced, but since the MSC-U isn't balanced output, and that would be shorting to a sort of floating ground which is part of the circuit, after two transformers rather than one, that gets weird real fast. But since the MSC-U is essentially a balun-un (balanced to two unbalanced signals) and since the RDL D.A. can output unbalanced, whether you use or ignore the MSC-U ring output won't hurt anything at the RDL D.A. I'm not sure if the MSC-U does any real common mode noise rejection (their tech support would reply sometimes, but wouldn't answer that either).

With levels dialed in, I still seemed to notice a disparity when the actual pulpit mic was hooked up. I'm not sure what levels it actually outputs because it depends on the pressure that hits it, and I can't access the DSP settings to see what they expect as input. Thus I tried to send in a talk from President Nelson, from my phone's speaker, into the mic (with the pulpit mic volume on the QSC touchscreen at the Bishop's pedestal set to max), and I tried sending some Bach through the same way, and my multimeter levels were again bouncing (fancy multimeters can record the highest value found thus far), but when sending a sine wave through that way, the ear perceives volume differently based on frequency (and high frequency at high volume is especially painful). So in the end I just played those things loud into the pulpit mic (with its volume maxed), and set the RDL D.A. input gain according to the green and red leds (very minimal red flickering on sometimes), with its output set to the +4dbu marking. Then I measured the bouncing signal coming off of the TRS cable exiting the MSC-U output, from tip to sleeve (not ring) and determined that the highest I could see, that it ever bounced around to, was 0.45V at 600 ohms, which is -4.7dbu RMS. When differential balanced output from the RDL D.A. was +4dbu, output from MSC-U was -4.7dbu; the balanced phase inverted ring from the D.A. adds 6db headroom as compared to an unbalanced signal, so the unbalanced signal should have been -2dbu, thus the MSC-U drops 2.7db doing its impedance changing from 150 ohms differential balanced to 600 ohms for each unbalanced signal. Because some attenuators have 5% margin of error (many resistors do), 38dB attenuation is probably a good option, since it can take the -4.7dbu down to -42.7dbu, but could be off in either direction perhaps by 5%, so 1.9db error, thus it might be -40.8 to -44.6 dbu, and the highest mic level signal should be below -40dbu. If I needed a little bump beyond that, I could turn the RDL D.A. output dial a little lower.

So consider using the following (or equivalent):

(Select quantity of two; select 3ft. each [or up to about 20 ft. collectively for both depending on your needs]; the first plugs 3.5mm TRS male into the MSC-U output) ( https://www.showmecables.com/3-5mm-ster ... l-rca-plug )

(One; RCA female attaches to white RCA male on one of the cables; the red RCA on that cable is a dead end) ( https://www.showmecables.com/f-type-mal ... le-adapter )

(One; it's important that the frequency goes to DC or Zero Volts on each attenuator; it's also important that at least one attenuator is DC blocking; attaches to f-type male of preceding adapter) https://www.showmecables.com/f-type-mal ... eSIfxGlwHf

(One; attaches to preceding attenuator) https://www.showmecables.com/f-type-mal ... -mhz-12-db

(One; attaches to preceding attenuator) https://www.showmecables.com/f-type-mal ... 0-mhz-6-db

(One; attaches to preceding attenuator) https://www.showmecables.com/f-type-fem ... eSIfxGlwHf

(Then attach the second cable's white RCA male to the RCA female of the preceding adapter; the male red RCA is a dead end; the cable's 3.5mm male TRS goes into the Sabrent AU-MMSA pink mic port.)

(One; pick a color light enough to label/write on, like white; tape up all the connections and label it to indicate its purpose) https://www.showmecables.com/pvc-colore ... -ft-length

(One; our laptop had no mic boost setting in the OS with which to fiddle) https://www.amazon.com/Sabrent-External ... 00IRVQ0F8/

This setup should work for nearly every meetinghouse or Stake Center, and for most any Windows or Apple computer, with parts that can be put together easily by any Tech Specialist.

It worked for my meetinghouse, and Zoom Meetings' audio now sounds better than being live in the chapel, for persons speaking at the pulpit, and sounds good for organ music, and decent for congregation singing, relative to sound entering the pulpit mic, based on its pointed direction, distance, etc. Background noise isn't much of an issue. I use the 'Original Sound For Musicians' setting in Zoom, with 'Echo Cancellation.' Also our meetings get started from the 'Recurring Meetings' section at the bottom right in Zoom (click the camera icon), rather than starting a 'New Meeting,' or 'Joining.' We've had issues with our host computer ending up in a different meeting from the participants when we don't go through our recurring meeting section.

Best wishes!

[leftofdamascus]

My tests mentioned in my earlier post actually seem valid despite the mic pressure being variable: it needs a mic level signal, so sending a sine wave from a tone generator website on your cellphone into the crab box, with the volume dial in the tone generator, and for your phone, and at the bishop's pedestal, set to max, and dialing in the crab box volume dial to have its XLR output at -40dBu to -60dBu mic level (using a multimeter that can handle the frequency of the sine wave) appears valid, then you can send that signal into the pulpit mic XLR port (with the pulpit mic removed), and measure the output of the dsp that goes to the audio distribution amplifier that feeds your rec. out ports, etc., if you ever need to. My calculations were wrongly in dbm which is only the same as dbu at 600 ohms, so I had to recalculate my dB values.

My 400hz test and the 100hz test show a difference of ~40db in what I send into the dsp (-54dbu and -48dbu respectively) and what I get out (-13dbu & -8dbu respectively) of the dsp.
My 40hz test shows I was actually sending in -46dbu into the dsp and getting out of the dsp -30dbu (-31dbV), which is only a 16db difference (thus presumably they have a high pass filter limiting low frequency signals, in their qsys design file for our dsp).

I did other tests with white noise where I could turn up the volume really quickly and it would let the signal through, and after about 4 seconds it would cut it off unless it was very loud. This seemed to confirm that our dsp is using an automatic gating mic mixer, that changes the noise floor dynamically, and would learn the white noise signal, treat it as slow changing rms, and filter it out, and raise the level it regards as noise floor, and thus the threshold over noise floor would require even greater volume to be satisfied. Long story shorter, when people speak into our pulpit mic, the signal is strong enough to overcome the threshold over noise floor setting, and when the pulpit mic needs to pick up music and congregational singing, it would only intermittently do so, because the threshold over noise floor setting was set too high to allow our pulpit mic to pick up organ music and congregational singing reliably. Our stake building representative, FM Group, contact at church HQ, audio blueprint guy, audio installers, and meetinghouse technology coordination team, each decided not to bother helping that setting get changed, or helping with a workaround. I suppose none of them sustained the lowly tech specialist in their calling, in a meeting, and thus they didn't when their help was requested. To the new techs, be prepared to battle all heck by yourself. That is as good of advice as you'll find on this forum.

In the end I was able to send our organ to the EJ-8 crab box, and into an XLR port into the dsp, to get the organ music. For congregational singing, I recommended using a Shure B.G. 2.0 older hand mic on the flat tabletop of the podium, pointed at the audience, and because it is cardoid, not hypercardoid, it avoids comb filtering issues with the pulpit mic because it doesn't also pick up a signal from its rear which is in parallel with the pulpit mic. This hand mic goes into a mini mic stand, connecting to an XLR cable which goes into an old Shure M68FC mic mixer (from ~1984), and out its line level RCA output (because mic level signals couldn't get over our noise gate in the dsp [threshold over noise floor], except for the organ feed, because we are reducing it to mic level from line level, so I can send it in at a very hot mic level). The mixer RCA output is high impedance, so I couldn't send it into the Emtech MSC-W which doesn't list specs, and their tech support is dodgy on listing specs, answering questions, or product support beyond a year that isn't slated for a new sale; nevertheless, their EJ-10 uses 4.7kohm impedance on aux ports, so if it was anything like that, that was way too low. The Extron DTP T HWP 4K 331 D aux input could handle greater than 10kohm impedance but it wasn't hooked up. The audio blueprint installer refused to hook it up when I asked him to in a message, and in the building, stating church HQ didn't intend for its use (and thus it would remain a dead port which people will plug into for the next 20 years). So I recommended the Ward move the blue clip from the Extron HAE 100 4k plus (HDMI audio de-embedder) to the Extron DTP HDMI 330 Rx (both in the back of the A/V rack) for sacrament meeting (disabling HDMI input on the same device as the Extron aux in, but enabling the aux in), and that blue clip can be put back for other meetings that need HDMI IN (projector stuff) which don't need audio in the foyer, mother's lounge, overflows, Zoom Meetings, etc., or someone can be moving that clip during a meeting. The mixer volume dial for the hand mic gets set at '10' and the master volume dial gets set to the lowest volume that lets congregational singing be appropriately heard in the foyer (any louder can introduce background noise into the meeting), and if that is too quiet, the program audio button on the bishop's pedestal QSC lcd screen can be toggled on for music, off otherwise. It must be on to hear that hand mic either way, and its volume slider is relevant in mixing it against the pulpit mic.

So, the coax attenuator and dc block cable I mentioned earlier in this thread: it didn't fix my intermittent audio but still provides the signal the Sabrent AU-MMSA wants, at least prolonging its life. We did crash the computer one time presumably from sending a line level signal into that device and unplugging the Sabrent device (as it had no eject USB option), so the higher voltage is a gamble if not building that cable. But, from what I understand about the Emtech MSC-U being a bal-un-un, when sending that signal into the Sabrent without the coax attenuators and dc block, with a plain old trs cable, I'd suspect that the Sabrent tip (left) channel sees the close side at line level, but its tip (left) channel far side at something higher than ground (higher than zero volts) because the stereo port ring (right) channel is probably actually hooked up (Sabrent wasn't sure if it is because it is used as stereo in other Sabrent devices, but not intended to be used as stereo in this device), so then you have multiple resistors, inverted signal, etc, running into the ground side (the far side of the tip

channel), and thus perhaps that does some attenuation of a sort, when those two sides for the left channel are compared. So, it might work without the coax attenuators and dc block, and apparently does, but it is at least specification-wise, sketchy.

Notes_251222_135528_e80.png

(https://electronics.stackexchange.com/a/253550)

"There isn't really a problem. As long as the return conductor resistance is low relative to the resistance / impedance of the speakers then the speakers lower terminals will be held at GND." -Transistor

In this instance, I'm not sure the return conductor resistance is held at ground (zero volts) or the Sabrent would see line level and it would be crazy distorted and way too hot. The PC we used didn't even have a 'Mic Boost' setting.

But all in all, this device worked for us. Thanks! Sorry for the tangent.

[davidanderspatten]

I've tried a number of the items listed (Headset Buddy and others) with similar results. What worked best for me was the "Behringer U-PHORIA UM2 Audiophile 2x2 USB Audio Interface" at about $45 on Amazon (https://www.amazon.com/dp/B00EK1OTZC).

[flarg2]

Our meeting house is equipped with the “listen” brand assisted listening devices. we just plugged one of those into our laptop via a USB to mic/headphone adapter and a 3.5mm male/male cable. Only hiccup we’ve had with this is we have to keep an ear piece plugged into the listen unit as well because it requires a load to stay powered up.

It’s a bit of a twisty solution, but it allows us to have the streaming laptop at the rear of the chapel near our bootleg camera without digging into the building wiring.

[russellhltn]

Only hiccup we’ve had with this is we have to keep an ear piece plugged into the listen unit as well because it requires a load to stay powered up.

I know that was true of the Comtek brand, but I didn't think the Listen came that way by default. It can be changed via programming.

The problem with having an ear piece plugged in is that it will act as a microphone and color the sound. It would be better to have a resistor of the same resistance.