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Help with basic limiter/VCA at end of diy synth signal path
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floating-water



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PostPosted: Tue Oct 06, 2020 7:20 pm    Post subject: Help with basic limiter/VCA at end of diy synth signal path Reply with quote  Mark this post and the followings unread

I'm having some trouble filtering through exactly what it is that I need on google and am on a time constraint with this project so if forgive my ignorance only been doing electronics for a few months this is just for an art exam

Could anyone suggest some kind of limiting circuit/other ideas I could put at the end of the signal path of a drone type synth. It puts out noise at a stupid level due to the fuzz-delay circuit at the end which is fine but it would take your head off with headphones straight into the output jack which someone might do. It runs off 9v. Ive heard vactrols can work? Anything passive would be ideal

Any help would be appreciated, thanks
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Grumble



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PostPosted: Wed Oct 07, 2020 11:01 pm    Post subject: Reply with quote  Mark this post and the followings unread

You could use a resistor in series with two zener diodes, which on its turn are in series with the anodes connected to eachother.
Pick your signal from where the resistor is connected to the zeners.
Choose the apropiate values for your application.


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floating-water



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PostPosted: Thu Oct 08, 2020 11:25 am    Post subject: Reply with quote  Mark this post and the followings unread

Beautiful thank you, so is that input to the labelled resistor, output from the diodes? What do they do, it looks like an overdrive/distortion type concept do the clip the signal?

Edit: i found some information on this type of circuit but im confused where the input/output is. When 'voltage in' and 'voltage out' are stated they're referring to the audio signal? In my case? In the diagram you show, thats a function generator right? So those two lines are two different signals?
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Grumble



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PostPosted: Thu Oct 08, 2020 2:30 pm    Post subject: Reply with quote  Mark this post and the followings unread

Voltage in is to the left, output is over the resistor to the right while the resistor most right might be the input impedance of the module following this circuit.
Mind you: the voltage over a conducting diode Is approx. 0.7 volts add this to the reverse voltage across the zener diode and you have the maximum (ish) voltage you need.
For example: If the zener voltage of both zeners is 3.3 volt you get a maximum voltage of 4volt positive to 4volt negative.

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floating-water



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PostPosted: Thu Oct 08, 2020 2:45 pm    Post subject: Reply with quote  Mark this post and the followings unread

OK thanks mr grumble, so when we're talking about volume in an audio generating analog circuit does that also equate to voltage? So a high volume would be a high voltage? Hence we're using diodes to set the maximum voltage/volume?

Also when you say 'voltage of both zeners' would that be all together, not individually?
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PostPosted: Fri Oct 09, 2020 4:00 am    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
OK thanks mr grumble, so when we're talking about volume in an audio generating analog circuit does that also equate to voltage? So a high volume would be a high voltage? Hence we're using diodes to set the maximum voltage/volume?
Exactly 🙂👍🏻

floating-water wrote:
Also when you say 'voltage of both zeners' would that be all together, not individually?

Since the zeners are connected anti-serial (is that even English?) one is conducting and the other is in reverse (which means that the max voltage is the zener voltage) and the output voltage is across both zeners, the answer is : Yes

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floating-water



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PostPosted: Sat Oct 10, 2020 5:04 am    Post subject: Reply with quote  Mark this post and the followings unread

OK, one last thing, your example of 'zener voltage of both zeners is 3.3 volt you get a maximum voltage of 4volt positive to 4volt negative' - does that mean there's 8 volts across/output? I'm still confused by the negative not just meaning ground in synth designs. And is there a chart for equating voltage to volume/db or is that not the right way to think of it

Thanks
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PostPosted: Sat Oct 10, 2020 7:31 am    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
I'm still confused by the negative not just meaning ground in synth designs

If a voltage is negative or positive depends on what you use as a reference. If you take a single 9V battery you only have 2 terminals to connect
to so you will get a single voltage of 9V. However the polarity of this voltage depends on what you use as your reference. Usually this will be the
(-) terminal so compared to that the other terminal is 9V higher and that gives you a positive voltage or +9V. But you could use the positive (+)
terminal as your reference too. In that case the other terminal is 9V lower than your reference in other words it's a negative voltage or -9V.
In most cases the (-) will be your reference/GND so you will have a single positive supply voltage.

Now consider putting two 9V batteries in series, so connect the (+) terminal of one battery to the (-) terminal of the other and let's label this point (x).
So you will end up with 3 terminals: (-) on one battery, (x) the connection between the batteries and (+) on the other battery.
If you measure between (-) and (+) you would see a voltage of 18V. And depending on how you measure it, it will be either -18V or +18V.
But you can also use (x) as your reference point. This way you will measure -9V between (x) & (-) and +9V between (x) & (+). This is what's
referred to as a bipolar supply and it's used in most synth designs where (x) will be your GND connection. Reason for using a bipolar supply
in synths is because you are mostly working with AC signals which need a reference voltage somewhere between the highest and lowest levels.
It's most convenient to have this reference at the same level for all circuits you use (especially with a modular synth) and it also has to be very
stable otherwise you are going to have a bad time. So that's where a bipolar supply is useful as it has a very stable 0V reference.

You can emulate a bipolar supply with a single supply by adding a fake ground, sometimes referred to as a virtual ground but this is a term for
something opamp specific so it's better to not call it that to avoid any confusion. It's not as stable as a real ground but for simple circuits it will suffice.
This is a technique often used in guitar pedals to make power distribution a lot easier. And this also gets us to your first question.


floating-water wrote:
OK, one last thing, your example of 'zener voltage of both zeners is 3.3 volt you get a maximum voltage of 4volt positive to 4volt negative' - does that mean there's 8 volts across/output?

It could be 8 Volt across the output but it depends a bit on your signal. If it is a bipolar signal, meaning it has a positive and a negative voltage
(referenced to ground), then yes the output swing would be clipped at maximum 8Vpp (peak to peak). Most audio circuits do have a bipolar signal
and for a single supply circuit this usually means there is a capacitor on the output, which kind of shifts the whole signal so it is referenced to GND.
For example if you have a circuit that produces a signal that fluctuates between 0v(GND) and +9V then after the output capacitor it could now
fluctuate between -4.5V and +4.5V. This does depend on what you reference the other side of the capacitor to but this will usually be GND.
It doesn't work perfect though but in general that's what you'll see with single supply circuits. You will probably come across circuits that label
this capacitor as DC-blocking or AC-coupling and it is also used on inputs.

Now if you would not have this capacitor and the output would indeed fluctuate between 0V and +9V then the zener diodes will only clip the
positive half of the signal so it would be clipped at +4V making the total amplitude 4Vpp. (in that case you would actually only need 1 diode).


Quote:
And is there a chart for equating voltage to volume/db or is that not the right way to think of it

There is, although personally I find it rather confusing to use dB's. When I think of dB I think of the actual audible loudness/pressure of a signal
for example coming from a set of speakers. Hower there is no direct relation between voltage and dB's for all speakers as it depends a lot on the
design most importantly the SPL factor. Now for line level signals (which is what you probably need) levels are often labeled in dB(V) and corresponding
voltages, here's a chart from wikipedia showing different line level standards:

Posted Image, might have been reduced in size. Click Image to view fullscreen.


I do wonder if clipping with diodes is what you are after but I will make a seperate post for that.

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PostPosted: Sat Oct 10, 2020 8:01 am    Post subject: Re: Help with basic limiter/VCA at end of diy synth signal path Reply with quote  Mark this post and the followings unread

continuing from my last post
Quote:
I do wonder if clipping with diodes is what you are after but I will make a seperate post for that.

Although diodes will definitely limit the signal they're not very kind to them and just chop off whatever is above their limit. For a squarewave
or a DC signal this is fine and it will mostly just reduce the amplitude but for any other signal it will affect the waveshape. For example a triangle
wave will turn into a trapezoid looking wave and you can say goodbye to those nice curves a sinewave has which in the worse case will end up
resembling a squarewave. Of course different waveshapes also sound very different so if you clip it with some diodes what you get out of it is
not just the same as what you put into it but at a lower level. This doesn't have to be a bad thing though, a lot of distortion/fuzz effects make use
of clipping the signal with diodes to create extra harmonics by changing the waveshape which will result in a richer sound.

I don't think that is what you are looking for though and you might just need a simple attenuator, which does bring the level down but without
affecting the waveshape. Of course this will only work if your signal has a somewhat constant level to begin with otherwise you would need some
kind of automatic gain control (AGC) which would keep the overall level constant but that is a lot more complicated.

For an attenuator you could use a potentiometer so the level can be adjusted or just 2 fixed resitors to bring it down to a more usuable level.
You can also combine a resistor and a potentiometer to have an adjustable output level but with a limited maximum. For headphones this might
also work but it would be better to buffer the signal which you sadly have to do in an active way. Reason being that headphones have a rather low
impedance and need a bit of current to properly work but a passive attenuator will limit this current. The same is true for the diode clipping btw
in which the series resistor is also limiting the current.


What I wonder about is if it could be done easier by adjusting the fuzz-delay circuit itself. It might be as simple as changing a single resistor but for
that I need to see a circuit diagram.

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floating-water



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PostPosted: Sat Oct 10, 2020 10:21 am    Post subject: Reply with quote  Mark this post and the followings unread

Damn son that was a read. I took in what I could but i think youre right the fuzz delay might need tweaking first its out of control and im probably being lazy. I have been trying to figure out how to tame the volume on it maybe you could help and then I can figure whats needed at the end once its sorted.
Its basically just the noise ensemble circuit:

Posted Image, might have been reduced in size. Click Image to view fullscreen.

at the moment its good volume with the pot at about 1/3 way through

edit: im actually looking to replace the vol pot with a resistor if possible, im using a 100k pot at the moment but im unsure how to do it properly, thanks
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PostPosted: Sat Oct 10, 2020 4:34 pm    Post subject: Reply with quote  Mark this post and the followings unread

I drew the schematic based on that layout which has some interesting things.

C4 and C5 are the AC-coupling capacitors for in-/output.
As you can see C4 is connected to pin2 of the PT2399 with R2. Pin2 is the internal reference voltage created
by the PT2399, basically a fake ground, so the input signal is shifted up so that it fluctuates above and below that
voltage. On the output the opposite happens with C5 in combination with P2 which shifts the voltage back to
having it referenced to GND.

R4,R5,R6 set the dry/wet mix and the output level.
if R4 is smaller than R5 (or R5 larger than R4) you will get more of the dry signal
if R5 is smaller than R4 (or R4 larger than R5) you will get more of the wet signal
R6 in combination with R4/R5 sets the output level of the mix. So if you make R6 smaller (or R4/R5 larger)
it should reduce the output level.

R3 seems to be the main feature of this circuit (apart from just being a delay) as it feeds the input signal into
the VCO of the internal delay clock. This means that the input signal has an effect on the delay time. It might
be interesting to replace R3 with a potentiometer or put it in series with R3 (and make R3 smaller).


note that the ouput from P2 is not really suitable to drive a pair of headphones directly.


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floating-water



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PostPosted: Sat Oct 10, 2020 5:58 pm    Post subject: Reply with quote  Mark this post and the followings unread

Mate youre a legend, took me ages to breadboard from verolayout schematic so much clearer. Ye I've got a basic lm386 amp driving everything to output

It goes 6 transistor oscillators into noise ensemble into filter into lm386.

Im gonna have read over all the circuit notes once i got some sleep
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floating-water



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PostPosted: Sat Oct 10, 2020 8:50 pm    Post subject: Reply with quote  Mark this post and the followings unread

You were right, although i never doubted it Let us pray but i brought R6 down to 500r or something like that cant remember, brought the volume right down even when the circuit is pushed its not violent on headphones. Im assuming its because the circuit is meant for guitars abysmal output?

How would I go about making the vol potentiometer fixed? My brain is saying a resistor across ground and out of c5, does output just become c5?

edit: another thing im not having much luck with the classic 'starve' knob, im running it off the mains and it works 9v or 12v but ive tried all the information i can find online, its works 'best' with 10k lin and a resistor across the pot but do you have insight on a right way to do it? At the moment it just basically turns 'on' when the pot is at full instead of a smooth gradient of voltage
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PostPosted: Sat Oct 10, 2020 11:41 pm    Post subject: Reply with quote  Mark this post and the followings unread

I was under the impression you needed some kind of passive emergency limiter to make sure the module where this was intended for wouldn’t destroy the input circuit of the following module.
All signals below the diode voltages would be undistorted, but will clip (and thus distort) when the voltages are higher than the diode voltages.

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PostPosted: Sun Oct 11, 2020 12:08 pm    Post subject: Reply with quote  Mark this post and the followings unread

I did yea but I hadn't tried tweaking the circuit that was causing the problem that required the passive limiter. I'm still looking to employ the limiter or at least learn about it for the future
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floating-water



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PostPosted: Sun Oct 11, 2020 12:27 pm    Post subject: Reply with quote  Mark this post and the followings unread

Grumble wrote:
I was under the impression you needed some kind of passive emergency limiter to make sure the module where this was intended for wouldn’t destroy the input circuit of the following module.
All signals below the diode voltages would be undistorted, but will clip (and thus distort) when the voltages are higher than the diode voltages.


Im still waiting to get some zeners online to test, ive only got a bunch schkotty kit diodes.
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PostPosted: Sun Oct 11, 2020 2:04 pm    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
You were right, although i never doubted it Let us pray but i brought R6 down to 500r or something like that cant remember, brought the volume right down even when the circuit is pushed its not violent on headphones. Im assuming its because the circuit is meant for guitars abysmal output?


good to hear that worked Very Happy

Personally I wouldn't go as low as 500 ohms (if that is indeed what you used) but make everything a factor 10 larger:
R4,R5 = 100K
R6 = 5K

This will reduce the current though the PT2399 might not have any problems with it, if it works it works.
But if you do want to make it battery powered it's always good to keep the current low.
It might affect the sound, maybe you'll like it better maybe you don't so just experiment with it.

Judging by the 2.2M resistor on the input (I'd reduce that to 10K..100K) it does indeed look like it's designed for a guitar.
The gain doesn't seem very high with those resistor values but the PT2399 might have some more internal gain.
I do wonder if reducing the output signal is the best idea, generally you want the signal in a chain to be as high as possible
untill the end for a better signal to noise ratio. Of course I don't know how high the signal from the PT2399 actually is.
I guess that also depends on if you want the ouput to be available or if it is only hardwired to the next circuit.


Quote:
How would I go about making the vol potentiometer fixed? My brain is saying a resistor across ground and out of c5, does output just become c5?

If you want to have the signal as if the pot was turned all the way up then yes. Otherwise you'd have to replace the pot with 2 resistors.
You might not need C5 or a resistor though depending on what is happening next with the signal. You mentioned it goes into a filter
before going into a LM386. This filter might have an AC coupling cap on the input already and you probably don't need to put 2 in series.
Never been a big fan of the LM386. It's a very convenient little chip but prone to oscillations and the standard gain, which can only be
increased externally, is too high for a lot of applications.


Quote:
edit: another thing im not having much luck with the classic 'starve' knob, im running it off the mains and it works 9v or 12v but ive tried all the information i can find online, its works 'best' with 10k lin and a resistor across the pot but do you have insight on a right way to do it? At the moment it just basically turns 'on' when the pot is at full instead of a smooth gradient of voltage

Success rate depends on what you try to starve. Something like a PT2399 might just shut down below a certain point.
Note that you would have to connect the starve pot after the 7805


btw if you want to experiment with diode limiting you can also use 2 LEDs in antiparallel (that is english).

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floating-water



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PostPosted: Mon Oct 12, 2020 3:41 am    Post subject: Reply with quote  Mark this post and the followings unread

Well the filter is just an RC filter for now, as basic as it gets, this is just 1st version of this synth.

Thanks for the tips about the resistors, ill have a play around.

So how would you recommed wiring up the starve knob, if the oscillators are first in the circuit I guess I could just have two wires coming of the power jack? One going to the starve knob ->osc, one going to the pt99 ->amp?
What other amp ic's would you recommend for this application? There was such an abundance of info on the lm386 i just went with it
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PostPosted: Mon Oct 12, 2020 6:26 am    Post subject: Reply with quote  Mark this post and the followings unread

I'm something of an ignoramus when it comes to compressors, so forgive me if this is a dumb question, but would a rack compressor do the job here? Are there hard limit compressors? (I don't know the words needed... ugh...)
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PostPosted: Mon Oct 12, 2020 7:34 am    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
Well the filter is just an RC filter for now, as basic as it gets, this is just 1st version of this synth.

is it adjustable or does it have a fixed frequency ?
Also is it a lowpass or highpass filter and what values did you use ?
it might be possible to integrate it into the noise ensemble circuit.

Quote:
So how would you recommed wiring up the starve knob, if the oscillators are first in the circuit I guess I could just have two wires coming of the power jack? One going to the starve knob ->osc, one going to the pt99 ->amp?

yeah, just split the power from where it comes in.
For starving it sometimes has a nice effect if you add a small capacitor (experiment) to ground after the starve pot.
btw there are different ways of wiring up the starvepot, most common is to use it as a variable resistor in series with the power,
but you can also wire it up as a voltage divider. So the outer lugs to V+ and GND (might need a resistor in series) and then take the power
out from the wiper. The only difference is really that one end of the pot is connected to GND or not so you could add a switch there.

Quote:
What other amp ic's would you recommend for this application? There was such an abundance of info on the lm386 i just went with it

If you just want to be able to drive a pair of headphones and/or use it as a line output then an NE5532/NE5534 can work quite well like in this circuit:
http://www.hobby-circuits.com/circuits/audio/audio-amplifier/749/portable-9v-headphone-amplifier
You could also do it with a single transistor and you can find some examples here:
https://www.hackster.io/jwzumwalt/single-transistor-audio-amp-32033a

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PostPosted: Tue Oct 13, 2020 2:32 am    Post subject: Reply with quote  Mark this post and the followings unread

This is where my ignorance may start to frustrate you, with the filter I've l literally just put a cap on a 50k pot, how can I make this more efficient? Continue with the passive topology for now. However I noticed placing the 'filter' before the pt99 got some wild resultsl; washes of white noise and oscillation

So with the amplifying circuit im gonna stick with the 386 for now as i have a bunch of them at hand however is there some kind of summing thing I could implement for the output so if you plug in headphones with a stereo adaptor its sums the audio into both ears, is that a thing? Looking at that schem with the NE's its using two op-amps with what looks like two inputs and a stereo jack?

Also you teased me with an LED anti parallel thing before? What is that exactly? I did a search online but the info is cacophonous
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floating-water



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PostPosted: Tue Oct 13, 2020 9:31 am    Post subject: Reply with quote  Mark this post and the followings unread

Starve knob works perfectly now and also acts as a master pitch knob. Putting it before everything including the amplifier, therefore turning it off was probably the stupidest thing i've ever done Microsoft
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PostPosted: Tue Oct 13, 2020 4:34 pm    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
This is where my ignorance may start to frustrate you, with the filter I've l literally just put a cap on a 50k pot, how can I make this more efficient? Continue with the passive topology for now. However I noticed placing the 'filter' before the pt99 got some wild resultsl; washes of white noise and oscillation

you could have a look at this page for some filter topologies, but if you have something you like just keep it.
The advantage of integrating it into the noise ensemble would be that the signal is buffered so you don't get any further
loss in signal strength nor does it get affected by what you connect to it.
But adjustable filters are a bit more tricky to make active.

Quote:
So with the amplifying circuit im gonna stick with the 386 for now as i have a bunch of them at hand however is there some kind of summing thing I could implement for the output so if you plug in headphones with a stereo adaptor its sums the audio into both ears, is that a thing? Looking at that schem with the NE's its using two op-amps with what looks like two inputs and a stereo jack?

The NE5532 is amp is stereo so does indeed have a stereo input and 2 outputs. But you can simply send the same signal to both sides
of the headphones. This will place them in parallel but the impedance is usually high enough that it shouldn't be a problem.
But if you do this, don't plug in a mono jack as that will short it out. Or you could place 2 resistors in series with the outputs like I did
for the NANDulator output. (you need some lower value resistors for use with headphones)

Quote:
Also you teased me with an LED anti parallel thing before? What is that exactly? I did a search online but the info is cacophonous

It's the same as using 2 antiparallel diodes, so 1 diode between the signal and ground pointing one way and 1 diode pointing the other way with
a current limiting resistor in series. so like (a):
Posted Image, might have been reduced in size. Click Image to view fullscreen.

Standard silicon diodes have a forward voltage of about 0.7V so that is what it will limit the signal to in the same way as when using zenerdiodes.
LEDS are also diodes so you can use those too but they have a forward voltage somewhere between 2V and 3V depending on type and color so
the signal is clipped at a higher level. You can also put multiple diodes/LEDs in series to get a higher voltage and use different combinations for
asymmetrical clipping.
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floating-water



Joined: Oct 06, 2020
Posts: 40
Location: UK

PostPosted: Thu Oct 15, 2020 3:30 pm    Post subject: Reply with quote  Mark this post and the followings unread

Appreciate all your help mate. I've ordered some NE's and some diodes to start implementing some of the lessons.

Might be a bit cheeky and not sure if its required to start a new thread but im balls deep in another project which I wouldnt mind some guidance with. It's a basic solid state guitar amp (lm386 type) and I'm trying to figure where and how I would implement a reverb circuit.

Lets say the lm386 circuit is this:
https://robrobinette.com/images/Guitar/Ruby/Ruby_Guitar_Amp_Schematic.pdf

And the reverb module id use is this:
http://www.accutronicsreverb.com/main/?skin=sub01_05_1.html

I understand the reverb needs to go before the lm386 as its a power amp, driving the speaker. However wouldn't the 386 affect the reverb? Wouldnt it distort? The reverb module will need buffering and filtering on each side of it but would that be sufficient for 'guitar -> amp with reverb -> out'. Should there be a pre amp before the reverb? Is pre amp the right circuit? Should it be boost instead?

Thanks
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PHOBoS



Joined: Jan 14, 2010
Posts: 4999
Location: Moon Base
Audio files: 670

PostPosted: Sun Oct 18, 2020 10:17 am    Post subject: Reply with quote  Mark this post and the followings unread

floating-water wrote:
I understand the reverb needs to go before the lm386 as its a power amp, driving the speaker. However wouldn't the 386 affect the reverb? Wouldnt it distort? The reverb module will need buffering and filtering on each side of it but would that be sufficient for 'guitar -> amp with reverb -> out'. Should there be a pre amp before the reverb? Is pre amp the right circuit? Should it be boost instead?

Thanks


The 386 should not really have any effect on the sound apart from amplifying it.
In practise it will always color the sound somewhat but as long as the signal isn't too hot you should be ok.
Doesn't look like that reverb module needs anything else as it is already AC-coupled, has a high input impedance
and a low output impedance. Though some filtering might improve it.

According to that page the peak input voltage is 1.5V so you don't want to go above that, but at the same time keep it
as high as possible. A guitar has a rather low output signal so yes you would probably need a preamp.

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