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1V audio burst every OTHER pushbutton

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also, this is for live sound, so unfortunately i won't have the luxury of the studio environment to go back and fix problems... gotta have a real-time solution, and something which is super easy to set up and calibrate because the occasion is ten grand pianos and i get very little sound check time!
 
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i like this option because i don't have to burn another channel for each piano which already will have a low, a left, and a right channel. i actually built a mechanical prototype last night of what i need to accomplish, but it is very clunky and likely to fail. and electronic solution would be must easier i think...
 
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To duck the thump, you need to detect the damper pedal transition before it hits the strings with delay according to expected pedal velocity and distance between trigger and thump, depending on position of switch sensor.
 
Tony, thats right, so i imagine i'll have to use some combination of the physical placement of the magnets and the hall-effect sensor, as well as the delay feature, to get the precise and correct timing.
 
All valid concerns of course, so I tested this out already on my yamaha cl5 console and with an attack of zero on the compressor and a low-pass at 100Hz, the 10kHz tone can't be heard even when it is near clipping the mic pre.

With a corner freq of 100 Hz, only the bottom 11 keys will be within 1 dB of flat response.

dougy's circuit was what I was going to recommend. One addition, a series R-C lowpass filter before the pot to sine-ify the square wave. Setting the filter unusually low, like at 2.5 kHz, will do a much better job of attenuating the square wave harmonics and attenuate the max signal down to about 1.25 V p-p.

ak
 
Your lowpass filter is set at only 100Hz and the output is called music? The treble tone control on my sound system operates at about 800Hz and when it is turned down all the way then a piano and any other musical instrument sounds awful. Muffled like crazy.
 
ak, thats a great idea... i didn't realize that it was a square wave. do you mind redrawing the circuit for me the way you'd put that in there? i have no clue how to add such a thing to this circuit

btw i'm using a stereo pair of typical piano mics for the mids and highs. they sound great, down to about 100Hz... thats where this whole thing comes in, with a specially designed mic for picking up low frequencies - but which tends to pick up that thumping noise as well. even though the thumping is an inherent part of the instrument (you can experience it by placing your head right down next to the strings), people usually find this sound annoying and unnatural, so it needs to go. also, sometimes pop piano genres can utilize low frequencies way beyond what would be considered appropriate for classical music, and those last 11 notes need to be really solid in the subwoofers.
 
Ok, I see that you have three mics and will feed the high frequency beep into the preamp of the low frequency "low frequency vibrations" mic input. Then after the beep triggers the compressor, it and the lowpass filter will reduce the beep sound to nothing.
 
that's the plan! now i have to figure out how to build a prototype of this circuit. i have no technical friends of your kind who could help me out in real life, so that's why i'm on here... would somebody be able to compile a list of parts i'd need to get? i think if i had the exact parts and a circuit drawing, i know someone who could at least help me assemble it. i really appreciate everybody's help so far!
 
OK new idea everyone! Is there a way I can use the 48V Phantom Power from the mixer board to power this thing? Instead of a battery? The mic i'm using is dynamic and doesn't care whether the phantom is on or off...
 
Still not clear - Do you want to inject the tone burst into a wired microphone signal out at the instrument? If so, then an XLR-in, XLR out box could have the tone burst generator inside along with a phantom power extractor and an input for the pedal sensor. But injecting audio onto a balanced dynamic microphone pair is not a simple task, especially when the tone is 50 times greater than the mic signal you want to preserve with recording-studio quality. Are you sure you don't want to bring the tone burst into the console on its own dedicated input channel?

ak
 
Yeah it needs to be combined with the incoming mic signal... I've got ten pianos to mic in this fashion, and I can't spare 10 extra channels just for the compressor key in. So I don't know which mic I'll be using, but let's say it's a Shure SM58 just for now. That mic ignores phantom power, and it is usually fine to Y together with other mics without too much loss in quality. So how best do I combine the SM58 and the burst signals, running the burst device from the Phantom Power, and assuring that the burst is at least 30dB louder than the max mic signal when the piano is being played loudly?
 
Let's assume the 5 Vpp (volts peak-to-peak) signal is a sine wave instead of a square wave. 30 dB below that is 5 / 31.6 = 158 mV. Divide by 2 for peak volts, and divide by root 2 for RMS, and you get a maximum microphone signal of 56 mVrms to meet your conditions. So, what is the SM58 output when a piano is being played loudly? This goes back to post #25 about filtering off the high freq harmonics and attenuating the squarewave amplitude. Maybe no attenuation needed. AG probably has this in his head...

Changing the 4093 to a 40106 gets you six inverters instead of 4, and you need a 5th one to create the opposite audio phase for summing into a balanced signal pair. Adding a signal diode takes care of gating the oscillator.
 
In the Yamaha mixer manual I could not find a 100hz lowpass filter but I found a 100Hz highpass filter.
I forgot to look at the overload input level for the mic preamp.
 
I don't know all the electrical parameters, but I do know that the max level i'll ever need from the burst is 1.4V rms. So i figure if i have a pot that goes from zero to 1.4V rms, i'll just play with it until it outputs what I need...?
 
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Changing the 4093 to a 40106 gets you six inverters instead of 4, and you need a 5th one to create the opposite audio phase for summing into a balanced signal pair. Adding a signal diode takes care of gating the oscillator.
Two of the NAND gates in the circuit I posted aren't actually required.
 
Dougy, if it isn't too much trouble, would you mind re-sketching that diagram for me with the added 48V power usage, the low-pass filter necessary to sine-ify the tone, and the appropriate pinouts for summing into a balanced mic signal? Am I missing anything? I'd really appreciate it!!
 
Sure, it's not much trouble. I can draw you the circuit with balanced output, but I don't know how well this will add to your existing mic output (do you know its output impedance? Is it 600 ohms?). The output impedance of the circuit below changes with the volume setting (which may not be an issue at all).

Using cabled 48V may introduce hum. A 9V battery is better for this reason.

Note: my u and N might look similar. The caps are labelled with 100N, 100uF and 1uF
 

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thanks!! so is that extra bit of sketching the 48V option? I literally know nothing about what you just drew out, but it seems that the circuit is built around a 9V input either way, so I can use a 9V battery or have the 48V option drop it down to 9V for the remainder of the circuit anyway? And what would the max RMS voltage of the output be with the pot all the way up? (I am no doubt showing my absolute ignorance here...)
 
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