DIYstompboxes.com

Hey all,

I would need a detector that reliably detects a plucked string even with fast runs and when notes are played legato.
Any ideas are highly appreciated!  :)

So far I tried it with a fw rectifier -> low-pass-filter (to avoid ripple of the envelope) -> opamp buffer -> high pass filter (470 nF in series followed by 10k to Vref) to remove DC component of the envelope -> comparator

Works quite fine. With faster runs some notes are skipped though.  :(


Regards,

Markus

Oops, didn't search that properly the last time  :icon_redface: Always searched for "pick detector"  ;)

Found this one http://www.diystompboxes.com/smfforum/index.php?topic=9546.msg57393#msg57393 (http://www.diystompboxes.com/smfforum/index.php?topic=9546.msg57393#msg57393)

Any comments on the AMS-100 "pluck detector" ?

Regards,

Markus

Thats a different thing then you want, even though it is very cool and unusual.  That makes a voltage that gets larger the faster you play, rather then making a reliable pulse every time you hit a note.

It seems like its possible to get a pretty good fast trigger from guitar, even though I've honestly never played anything that could do it right.  Despite the challenges, it could rely on a few things, since it is guitar specific, rather than a general music device.  A new note played on clean guitar played with a pick is a quick increase in volume, every time.  If you could monitor whether the volume increases quickly, like you were doing with a comparator, but also monitor how much it got louder relative to where it was an instant before the note, you could create a kind of floating threshold control to weed out little increases in volume from finger noise or envelope ripple.  Hammer ons and finger picking might be forfeit, though.

A cheap and dirty way to do it is to make a pick out of a conductive material and run a wire from that to the pedal, so when it touches the strings it will ground something and make a pulse with a one shot type circuit.  That would probably be bulletproof, and very fast responding too.  might be a fun mad scientist thing to rig up...

Hey Processaurus,

thanks for your tips. A conductive pick detector for sure would be an interesting option!
Think I will first try it the conventional way though  ;)
Just browsed briefly through this "Device" issue and as far as I understood the AMS-100 has a pluck detector. Will have to read more thouroughly this eve to really understand how it works.

Markus

The pluck detector is what you might call a "macro event".  When you pick a note or chord, that initiates an approximately DC envelope voltage that corresponds to the vibrating string's amplitude over time.  The trigger extractor compares the amplitude of that signal to some preset threshold assumed to correspond to what would be there IF a string had been deliberately plucked (rather than simply brushed against or inadvertently vibrating in sympathy because the damn PA is too loud).  If the envelope exceeds that critical threshold, the unit/circuit puts out a single pulse of a specific voltage and of a specific duration.  That pulse corresponds to roughly what a traditional analog mono synth keyboard would have put out, and it is the "note event indicator" that starts all the other balls rolling, such as the envelope generators that make the sound fade in or the filters sweep.

The "pluck detector" circuit in DEVICE is a kind of "accumulator" in the sense that it allows these individual pulses to continue to charge up a cap.  The voltage on that cap becomes the sum total of the various pulses within any given period of time, minus whatever leakage occurs.  Think of it like a bank account.  The leakage is what your food, rent, utilities, incidentals cost, and the incoming pulses are the pay from all the pickup gigs you're able to get.  If you get a lot of work in a short period of time, then the incoming far exceeds the outgoing and your account statement is higher.  If the gigs are only sporadic, then the bank account rises momentarily when a cheque comes in, only to dribble out again because of those ongoing and unavoidable costs.

So, if you play faster, AND the picking is clean enough to generate a trigger pulse each time you pick a string, then those individual triggers are summed by the pluck follower to produce a control voltrol roughly proportional to the speed of your playing.  If you only pick a note now and then, or don't pick clean and hard enough to generate a new trigger for each of the notes you are producing from the guitar (e.g., by hammering on, or doing two-handed fretting), then very little will accumulate in the pluck follower.

That's why I call it a "macro event".  It corresponds to what is going on, in general.

One of the very best envelope followers and trigger extractors I've ever used is Harry Bissell's.  Exquisitely responsive, though a couple of notches up from most conventional envelope followers in terms of circuitry and parts count.  Worth it, though.  Definitely worth it.

Hey Mark,

thanks a lot for the clarification  :) (thanks to Processaurus too  ;) ).
Am I correct that I could simply just take the part that generates the trigger pulses for my "pluck detector"? Should be the part with the slow and fast filter I assume?
Are there any other circuits that would do the trick? I just need a pulse for each pluck. It should however be fast enough to work with faster note runs.

Thanks a lot in advance!


Markus
 

Maybe this will do what you want?

http://www.musicfromouterspace.com/analogsynth/signaltogate.html

~ Charlie

The "trick" in getting a decent, reliable, valid and usable trigger from a guitar string is that the circuit needs to be adaptable and context-sensitive.  What ought to count as another note/strum played will vary all the time.  Indeed, many of our natural playing styles involves converting one picked note to another or simply using another fret as a way of mimicking another lightly picked note.  In such instances, the previous note will note have died out completely, and the new note is not different enough in level that it would necessarily register as a plucked note.  Certainly its initial transient is not robust enough to stand up and out against things like normal finger gliss as you move your finger along the string from here to there.  So, the sort of circuit that a great trigger detector needs to be is one that can perform the same sort of decision the human ear/brain does, and recognize that a new note has been moved to based on the detected end of the last one.  That, of course, relies on pitch-tracking and comparisons between two adjacent pitches.  Traditional trigger extractors (including Ray Wilson's that Charlie linked to) only make use of overall amplitude information; if its a big enough shift in amplitude then it must be a new note.

Another way of solving the problem has been to split the pickup such that the range of what each separate coil has to pay attention to is reduced.  Assisting with this is the use of harmonics instead of fundamentals as a means of identifying new notes.  So, if I use the content above 2khz for my low E, and identify new notes based on some sort of coherent harmonic content, then I may be able to set my trigger threshold low enough that many of the various random sources of pickup noise will be safely ignored and ignorable.

Harry Bissell's envelope follower is nice because there is normally a tradeoff between speed of response and ripple.  The faster it can react to sudden transients, the more likely it is to also react to slight deviations in amplitude as the note dies out (including any beats produced by things like localized string wear from frets).  When that envelope voltage is used to control things, they can sound bad and almost distorted (certainly people's initial suspicion is that there is something distorting).  If you do things to reduce that ripple, it generally results in the follower not being able to respond very quickly.  Since the trigger extractor doesn't put out a trigger pulse until the envelope voltage rises above the critical threshold, you end up with trigger delay on the order of several milliseconds, which can often be enough to throw off your timing.  What his envelope follower accomplishes is that it reacts fast enough to permit trigger extraction at speeds that don't interfere with normal playing, while successfully managing the potential for ripple. Best of both worlds. I played Harry's guitar synth, made from a G-Vox divided pickup (very similar to GK-1/2/3) mounted on an Ibanez überstrat, and you never really felt like you were playing a synth.  You felt like you were playing a guitar.  THAT is a good envelope ollower and trigger extractor.

Charlie, Mark, thanks a lot for your help!!  :)

Some more questions  ;)

The "Bissell follower" sounds very promising!
IIRC it  works with a couple of parallel stages. Mark, do you remember how many stages were in that synth you played??

Regards,

Markus

 Hmmmm... I remember Morley had... something with a plectrum and a wire going to the giant chrome... whatever it was. I can't remember what it was called or what it did, but they have made many of their schematics available for download on their site.

Jeez, that ain't very helpful, is it? Sorry, it's all I have...

Hey Markus, what's the application that you want to use this for?

I think the "unnatural" response that Mark is talking about is something that a lot of folks who use guitar/bass synth-type pedals and effects have just come to accept. Even though tracking is typically marginal, you can also adapt your playing style to get musical sounds out of these kinds of effects.

~ Charlie

Quote from: Mark Hammer on March 07, 2007, 02:44:17 PM
The "trick" in getting a decent, reliable, valid and usable trigger from a guitar string is that the circuit needs to be adaptable and context-sensitive. ....... So, the sort of circuit that a great trigger detector needs to be is one that can perform the same sort of decision the human ear/brain does, and recognize that a new note has been moved to based on the detected end of the last one. 

Harry Bissell's envelope follower is nice because there is normally a tradeoff between speed of response and ripple.  The faster it can react to sudden transients, the more likely it is to also react to slight deviations in amplitude as the note dies out..... What his envelope follower accomplishes is that it reacts fast enough to permit trigger extraction at speeds that don't interfere with normal playing, while successfully managing the potential for ripple.
Best of both worlds. I played Harry's guitar synth, made from a G-Vox divided pickup (very similar to GK-1/2/3) mounted on an Ibanez überstrat, and you never really felt like THAT is a good envelope follower and trigger extractor.

Just for the H of it, i googled "pluck detector"-
came up with these.......

http://www.sweetwater.com/store/detail/Prophesy/
....a guitar processor with "patent-pending", "pluck detection"  ??   Hmmmmmmm(?)
            (Maybe a fancy envelope follower??)   (like multi-band??)    ;)   ;)
            (Maybe they patented Bissell's circuit?????)

and this----
http://www.dafx.ca/proceedings/papers/p_025.pdf

and this----
http://www.cnmat.berkeley.edu/ICMC2000/pdf/pickup.pdf

not really related, but interesting reading................. (yawn)

Now, go back to Device1-1, page 9.
re-look at IC4A,B,C,D.....look familiar? ???
can U say...self....adaptive....comparator? ???
It's even got a LED   :icon_exclaim:

U R confusing amplitude with frequency.
When U pluck, it's the amplitude.
If U want pitch tracking, it's the frequency.
The Envelope Follower is the simplist form of "pluck" detection.   One Pluck, please.  Only one.
To detect a pluck On Top of a sustained signal...now, that's what I'm talking about.
And, BTW, Markus want's this for a bass note @ 30HZ  :o
And, Mark: using a  hexaphonic pickup is "cheating"    :icon_biggrin:
(but, hey, that's what I use on my RolandReady Strat  ) 8)

stay adaptable  ;)
T

Quote from: moosapotamus on March 07, 2007, 09:47:26 PM
Hey Markus, what's the application that you want to use this for?

I think the "unnatural" response that Mark is talking about is something that a lot of folks who use guitar/bass synth-type pedals and effects have just come to accept. Even though tracking is typically marginal, you can also adapt your playing style to get musical sounds out of these kinds of effects.

~ Charlie

Basically, I need it it as a side chain for switching a PLL based thingy. So, ideally it should detect each pluck no matter if you play fast runs or play notes legato.

QuoteTo detect a pluck On Top of a sustained signal...now, that's what I'm talking about.

Exactly!  :)

QuoteAnd, BTW, Markus want's this for a bass note @ 30HZ

Actually, I almost gave up on the idea of using a PLL with Bass.  ;)
It's too slow. For guitar it's somehow OK.

QuoteNow, go back to Device1-1, page 9.
re-look at IC4A,B,C,D.....look familiar?
can U say...self....adaptive....comparator?

As far as I understood the IC4A,B,C,D  part is two lp filters, one has a slower attack than the other. The two filter outputs are sent to a comparator which - again as far as I understood - should   

Quotedetect a pluck On Top of a sustained signal...

Please correct me if I'm wrong.

Don't know if this or the "Bissell follower" would be better for my purpose...

Thanks a lot for your help!  :)

Markus

Quote from: markusw on March 08, 2007, 01:44:49 AM

Quote from: moosapotamus on March 07, 2007, 09:47:26 PM
Hey Markus, what's the application that you want to use this for?
~ Charlie

Basically, I need it it as a side chain for switching a PLL based thingy.
So, ideally it should detect each pluck no matter if you play fast runs or play notes legato.

QuoteTo detect a pluck On Top of a sustained signal...now, that's what I'm talking about.

Exactly!  :)

QuoteAnd, BTW, Markus want's this for a bass note @ 30HZ

Actually, I almost gave up on the idea of using a PLL with Bass.  ;)
It's too slow. For guitar it's somehow OK.

WHAT???   :icon_eek:

Quote from: toneman on March 08, 2007, 10:39:11 AM


WHAT???   :icon_eek:

Hey T,

what does the "WHAT???  :icon_eek:" refer to?

Maybe I'll give it a try with bass again once the whole mess works. At the moment I'm testing it with guitar.
Everything gets so damned slow, not only the PLL but also e.g. the envelope follower I'm using at the moment for triggering....

Markus

WHAT?  referred to this:
"Actually, I almost gave up on the idea of using a PLL with Bass.  Wink
It's too slow. For guitar it's somehow OK."

Just when I've been looking into buying a Ibanez 4string bass guitar   :(
Well, will still buy it.  Can't check em out until Fri or sunday.
Looking at 2 ibanez's.
then use 2 harmonizers.  1st shifts up one octave; 2nd shifts that up another octave.
;)

I see U actually said "almost"    :D

amplitude isn't slow.
it's frequency(determination) that's slow.
so the pluck thingie should be do-able
IMO

BTW, i've got some "node errrors" in the simulations(?)
LTC says it's "missing n010"....but even this changes.
Still, bottom line: it's "missing" one node, so sim won't run(?)
will email for some insight on how to find and correct.
afn
T
:)

QuoteJust when I've been looking into buying a Ibanez 4string bass guitar   
Well, will still buy it.  Can't check em out until Fri or sunday.
Looking at 2 ibanez's.

QuoteI see U actually said "almost"   

Yup, almost but not yet  ;)  Right now after some more tweaking it seems possible again.  :)
For bass you'll need some dry signal for the punch anyway. So a bit of latency could still be OK.

Quoteit's frequency(determination) that's slow.
so the pluck thingie should be do-able
IMO

Would be cool!!  8)

Strange error you got in LTC..
Maybe send me the file so I can check?

Regards,

Markus

Quote from: markusw on March 08, 2007, 01:44:49 AM
Basically, I need it it as a side chain for switching a PLL based thingy. So, ideally it should detect each pluck no matter if you play fast runs or play notes legato.

QuoteTo detect a pluck On Top of a sustained signal...now, that's what I'm talking about.

Exactly!  :)

have you considered throwing Slew detection at it ?! 

... worse case, in a legato run where the general "envelope" sizes don't change too much there will still be some increased slewing at the front of plucks and hammer-ons - in fact a double time differentiation might be in order ... it's guaranteed that slewing within a physical string based signal will be monotonically decreasing in time following "any" sort of attack - that's what would allow the triggering to work without confusion ... with clear plucks it would be even easier because of the increase in rate differences - ie., more amplitude at a given frequency = higher SR (dV/dT) ... you only need a differentiator or two, DC averaging and a threshold detector to make it happen

the question that would remain is whether/which type of other noises would compete with those "rates" ... just an idea

JC

Quote from: Eb7+9 on March 08, 2007, 03:08:03 PM
have you considered throwing Slew detection at it ?! 

... worse case, in a legato run where the general "envelope" sizes don't change too much there will still be some increased slewing at the front of plucks and hammer-ons - in fact a double time differentiation might be in order ... it's guaranteed that slewing within a physical string based signal will be monotonically decreasing in time following "any" sort of attack - that's what would allow the triggering to work without confusion ... with clear plucks it would be even easier because of the increase in rate differences - ie., more amplitude at a given frequency = higher SR (dV/dT) ... you only need a differentiator or two, DC averaging and a threshold detector to make it happen

the question that would remain is whether/which type of other noises would compete with those "rates" ... just an idea

JC

Thanks a lot for your tip!  :)
Sounds very logic! Only problem is I don't have an idea on slew detection. Will have to do some googling.
Basically, you mean to differentiate the guitar signal or the envelope once or twice ?   :icon_redface:

Regards,

Markus

Please forgive me but I still cannot grasp what you are trying to achieve.

Quote from: StephenGiles on March 08, 2007, 03:50:20 PM
Please forgive me but I still cannot grasp what you are trying to achieve.

I would like to reliably get a pulse each time I plug a string. It should also work for fast runs and if notes are played legato. Of course hammer-ons would be cool too  ;)
Actually, I don't need an envelope, only if it's part of the pluck detector.

Regards,

Markus

Without wishing to be too much of a contrarian, I can't see why someone would want a trigger pulse and NOT want an envelope voltage unless the intent was simply to apply envelope generators to fairly normal effects like a phaser, etc. and use those to process a fairly conventional guitar signal.  The thing about having an envelope signal at your disposal is that you can add it to whatever envelope generators you have.

FX-Tracker...

[EDIT]: found it... (http://www.backline-eng.com/) - impressive!

Quote from: Mark Hammer on March 08, 2007, 04:12:31 PM
Without wishing to be too much of a contrarian, I can't see why someone would want a trigger pulse and NOT want an envelope voltage unless the intent was simply to apply envelope generators to fairly normal effects like a phaser, etc. and use those to process a fairly conventional guitar signal.  The thing about having an envelope signal at your disposal is that you can add it to whatever envelope generators you have.

I just need the trigger pulses to switch a sample/hold circuit and a few other connections within a PLL-based harmonizer. Since I intend to feed the harmonizer signal as a carrier into a ring modulator there is no need for an envelope.
Plain, reliable pulses with every pluck would be more than great  ;)
So far the idea worked out pretty well but I'm not really happy with my current pluck detector.

Regards,

Markus

Quote from: puretube on March 08, 2007, 04:14:42 PM
FX-Tracker...

[EDIT]: found it... (http://www.backline-eng.com/) - impressive!

definitely impressive  :icon_eek:

I think the guy has been here once, a year or 2 ago...

Dare I say it, he could be too smart for here.  That's one heck of a product.  Of course, it could be too smart for most musicians, too. :icon_wink:

at that time it appeared, I was not at all interested, coz I thought it were some computer plug-in,
or "virtual" software thingy (from the look of the first image-renderings...

but a recent thread at HC opened my eyes  :icon_cool:

(BUT: it`s digital  :icon_eek:)


:icon_biggrin:

had not ever heard of this!!    :o

thanx!! for the link & info PT  :icon_exclaim:
gonna go back an watch the videos   :)

the BOSS RC-50 is supposed to be the ultimate looper.
I have an RC-20 and it's OK, but the hardwired switch for changing loop really sucks.
I've heard demos of the Boomerang also.

I'm looking to do the same as Markus--  some type of harmonizer into my RingModulator carrier input.
BTW. I googled for "slew detector"  and couldn't find a thing!   >:(

may all your loops be fruity!!   :P
afn
8)
T

 WOW! That's something completely different! I especially like the MIDI clock out (even though I haven't used MIDI in years). This is the kinda stuff that's been in my 'I wish it were possible' mental list for a long time. One of the coolest things I've seen in a long time! Waveform outs, jeez it goes on and on.

I wonder how effective it really is? Like the tracking and stuff... If it's REALLY good it might even work on bass...

I think if I was to try and make a 'pluck detector', I would try to utilize the fact that when a string is 'plucked' it is more or less percussive. [speculation]This means it has a really sharp rise time, probably not harmonically related to the string being 'plucked'[/speculation]. A sharp rise time can be differentiated into a (non-ideal) impulse. Or a pulse of finite time in the real world. This can be setup to trigger just about anything you want. There would still be some 'thresholding' but it would be more of how hard/purposely do you have to pluck the string in order for it to be detected than how 'loud' the signal is coming from the string.

That's how I would approach it anyway.

FWIW,

--john

QuoteA sharp rise time can be differentiated into a (non-ideal) impulse.

Hey John,
thanks for your help!  :)
My current pluck detector has one differentiation step (of the envelope). I think of adding a second one.
Is it what you mean??

Regards,

Markus

Quote from: markusw on March 08, 2007, 04:09:11 PM

Quote from: StephenGiles on March 08, 2007, 03:50:20 PM
Please forgive me but I still cannot grasp what you are trying to achieve.

I would like to reliably get a pulse each time I plug a string. It should also work for fast runs and if notes are played legato. Of course hammer-ons would be cool too  ;)
Actually, I don't need an envelope, only if it's part of the pluck detector.

Regards,

Markus

I see, I'll look at the Arp Avatar manual over the weekend for anything that may be of help.

Quote from: StephenGiles on March 09, 2007, 04:40:13 AM

I see, I'll look at the Arp Avatar manual over the weekend for anything that may be of help.

Cool!  8) Thanks a lot!!

Markus

Quote from: markusw on March 08, 2007, 04:31:23 PM

Quote from: puretube on March 08, 2007, 04:14:42 PM
FX-Tracker...

[EDIT]: found it... (http://www.backline-eng.com/) - impressive!

definitely impressive  :icon_eek:

I agree - impressive but hardly of use in a band!

I breadboarded the AMS-100 pluck detector yesterday evening (modified for a single 12V supply) and it works pretty nice. Definitely better than my home-made "fwr-envelope-follower-double differentiator" thingy. It's very sensitive and triggers quite nicely over the whole guitar range. Depending on the gain it still has a preference for the lower or higher frequency range but it defintely is a valid option. Will start tweaking the various pots now to see how they change the response.

Regards,

Markus

Just curious.  How did you evaluate the responsiveness of it?

Quote from: Mark Hammer on March 14, 2007, 12:44:07 PM
Just curious.  How did you evaluate the responsiveness of it?

Firstly, there is a 555 connected to the pluck dtector wired as a pulse stretcher that gives a "on signal" for 100-150 ms. This pulse drives an LED indicator which clearly flashes with each pluck.
Secondly, the pluck detector is connected to a 4046 PLL plus a sample/hold stage. The PLL only changes frequency if the pluck detector triggers.
Once I optimised the whole mess I'll post the schematics so we can discuss how to improve it (or whether to forget it)  ;)

Markus

Cool.  Worth waiting for patiently.

Here is a sound sample of the PLL circuit with the AMS-100 pluck detector (PD). The PLL is set to 2 or in the second half of the clip to 3 octaves up and fed as a carrier into my Ring Stinger clone.
The PD triggers a NE555 which is set to give a "on signal" for about 100 ms (don't know exactly yet) with each pluck.
So with each pluck the circuit goes for 100 ms to sample and then back to hold keeping the VCO freq constant until the next pluck. A LF398 samples the 4046'S PC2 out voltage.
In sample state the 4046's PC2 out is connected directly (via the loop filter) to the 4046's VCO-in. After 100 ms the LF398 goes to hold and two 4066's disconnect PC2-out to VCO-in and connect LF398-out to VCO-in. At the same time the comparator that during sampling feed's the squared guitar signal into the 4046 is disconnected by another 4066 and the VCO is connected with the PLL's input. Therefore, after going back to hold  PC2-out remains at the level of the last note played.
In my hands this trick speeds up the lock-in or at least makes the glitch at the beginning of each note a bit less noticeable.
Interestingly, the PLL locks in faster when the strings are plucked a bit harder although the guitar signal squarer works at much lower levels. Also performance is better when the strings are plucked with a pick. Since I never used a pick with bass the clip is also done without. It's a cheap Strat copy at neck pu and recording was done directly into the PC w/o cab sim.

Let me know what you think  ;)

http://www.aronnelson.com/DIYFiles/up/Ring_Mod_PLL_SH_2_3_octaves_up_14-03-07.mp3 (http://www.aronnelson.com/DIYFiles/up/Ring_Mod_PLL_SH_2_3_octaves_up_14-03-07.mp3)

Markus

I don't understand the electronics of it but Penfold uses one a lot for his circuits in the Babani books based on an LM13700/13600 for triggering autowahs, tremelos etc. Might be of some use.

Quote from: col on March 15, 2007, 08:29:14 AM
I don't understand the electronics of it but Penfold uses one a lot for his circuits in the Babani books based on an LM13700/13600 for triggering autowahs, tremelos etc. Might be of some use.

Thanks!  :)
I'll have a look...

Markus

Quote from: markusw on March 14, 2007, 05:46:12 PM
Interestingly, the PLL locks in faster when the strings are plucked a bit harder although the guitar signal squarer works at much lower levels. Also performance is better when the strings are plucked with a pick. Since I never used a pick with bass the clip is also done without. It's a cheap Strat copy at neck pu and recording was done directly into the PC w/o cab sim.

Presumably the squarer is just an op-amp comparator?  If so, try feeding its output to the input of a schmidtt trigger Logic chip (40106 or similar) to clean it up, or - Unless you're likely to use the other devices in the package - you can even wire up a CMOS555 as a schmidtt trigger (takes up less space on a PCB).

Quote from: gez on March 15, 2007, 12:40:31 PM
Presumably the squarer is just an op-amp comparator?  If so, try feeding its output to the input of a schmidtt trigger Logic chip (40106 or similar) to clean it up, or - Unless you're likely to use the other devices in the package - you can even wire up a CMOS555 as a schmidtt trigger (takes up less space on a PCB).

Hey Gez,

thanks for the tip! It's a LM311 followed by a 4070 XOR gate (because I have it on breadboard). Do you think the 4070 is OK or would you use another one to clean it up?
The XOR is connected with one input directly to the LM311's out and with the other input to Vref. I also have some spare 4049 stages on breadboard. Would you prefer them over the 4070?

Markus

Quote from: markusw on March 15, 2007, 12:52:36 PM

Quote from: gez on March 15, 2007, 12:40:31 PM
Presumably the squarer is just an op-amp comparator?  If so, try feeding its output to the input of a schmidtt trigger Logic chip (40106 or similar) to clean it up, or - Unless you're likely to use the other devices in the package - you can even wire up a CMOS555 as a schmidtt trigger (takes up less space on a PCB).

Hey Gez,

thanks for the tip! It's a LM311 followed by a 4070 XOR gate (because I have it on breadboard). Do you think the 4070 is OK or would you use another one to clean it up?
The XOR is connected with one input directly to the LM311's out and with the other input to Vref. I also have some spare 4049 stages on breadboard. Would you prefer them over the 4070?

Markus

My experience of constructing oscillators using the 4070 (or using them as buffers) then trying to use their outputs to clock flip-flops says that they're not such a good choice.  I still say use a dedicated Schmidtt device - they're designed to give clean switching at 'fast' speeds (an audio signal isn't that fast) and I've found that it really makes a difference when driving PLLs (lock-on time is much quicker).  Also helps if you square the crap out of the signal before going to your comparator.  Diode clipper circuits with plenty of gain are ideal for this.

Hey Gez,

will give ot a try. Would one ore to 4049 stages be OK??

Here btw is a draft schem: http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch.pdf (http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch.pdf)
The input stage, lp filter and the first part of the AMS-100 PD is not shown. All opamps are TL072,4.
U8 and U9 simulate a LF398

Markus
.

Quote from: StephenGiles on March 09, 2007, 04:40:13 AM

Quote from: markusw on March 08, 2007, 04:09:11 PM

Quote from: StephenGiles on March 08, 2007, 03:50:20 PM
Please forgive me but I still cannot grasp what you are trying to achieve.

I would like to reliably get a pulse each time I plug a string. It should also work for fast runs and if notes are played legato. Of course hammer-ons would be cool too  ;)
Actually, I don't need an envelope, only if it's part of the pluck detector.

Regards,

Markus

I see, I'll look at the Arp Avatar manual over the weekend for anything that may be of help.

Couldn't see anythng of interest.

Quote from: markusw on March 15, 2007, 01:47:47 PM
Hey Gez,

will give ot a try. Would one ore to 4049 stages be OK??

Here btw is a draft schem: http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch.pdf (http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch.pdf)
The input stage, lp filter and the first part of the AMS-100 PD is not shown. All opamps are TL072,4.
U8 and U9 simulate a LF398

Markus
.

Haven't looked thru your preliminary schematic yet Markus, but look forward to perusing it at my leisure (probably over the weekend).

My preference when it comes to Schmidtt devices is the 40106 or 4093, though if you wire pins 2&6 together of a 555 and use pin 3 as the output it takes up less space and works just as well.

4049?  Never used them for Schmidtts.  If I were to use them, I'd wire two in series so that the output is non-inverting, then I'd wire up a resistor (100k say) from the output of the squarer to the input of the first inverter, and then another resistor (equal in value) from the output of the second inverter to the input of the first.  It's a common arrangement and you should be able to find a schematic on-line for it, though I have no idea how well it will perform. 

Although one of the inputs of a 4046 (haven't checked, but presumably you're using it?) can accept signals that are less than perfect, the cleaner the signal to this input the better the PLL behaves: less work for it to do (as it were).  For this reason, I've found that logic buffers with schmidtt inputs really make a difference in terms of tracking etc.

Hey Gez,

thanks a lot for your explanations!  :) Will give it a try for sure!

I found two bugs in the schemo. The file is updated.

Markus

One more Q: is the propagation delay the parameter that determines whether a device is suitable for cleaning up the signal?

QuoteAlthough one of the inputs of a 4046 (haven't checked, but presumably you're using it?) can accept signals that are less than perfect,

Forgot before: I'm using phase comp 2 which is supposed to be less immune to noise than PC1 but it can't lock in on harmonics.

Edit: I found a 4093  :)

Markus

Quote from: StephenGiles on March 15, 2007, 02:30:57 PM

Quote from: StephenGiles on March 09, 2007, 04:40:13 AM

Quote from: markusw on March 08, 2007, 04:09:11 PM

Quote from: StephenGiles on March 08, 2007, 03:50:20 PM
Please forgive me but I still cannot grasp what you are trying to achieve.

I would like to reliably get a pulse each time I plug a string. It should also work for fast runs and if notes are played legato. Of course hammer-ons would be cool too  ;)
Actually, I don't need an envelope, only if it's part of the pluck detector.

Regards,

Markus

I see, I'll look at the Arp Avatar manual over the weekend for anything that may be of help.

Couldn't see anythng of interest.

Nevertheless, thanks!!  :)

Quote from: markusw on March 15, 2007, 04:23:45 PM
One more Q: is the propagation delay the parameter that determines whether a device is suitable for cleaning up the signal?

More important (IMO) is the following:

http://en.wikipedia.org/wiki/Transition_time

All helps though.

It's interesting that you have both inputs of the LM311 biased to Vref.  I do something similar quite often using CMOS output amps and in their quiescent state, the outputs actually try to bias to half supply - though they're far from stable and swing all over the place - and the chips have a tendency to act as an amplifiers towards the tail end of notes, rather than comparators.  Because there's no negative feedback, the signal is always clipped, but the sides lean inwards as a signal decays.  This gives the effect of introducing dynamics - not in terms of amplitude but 'timbral' dynamics.  The end result makes for a more interesting comparator fuzz, but the signal needs cleaning up to be suitable for, say, driving the clock input of a flip-flop, which is why I use Schmidtt trigger buffers.

Do you own a scope?  I'd love to know if something similar happens with these chips (just play a note and watch the slope of the square output as the note decays).  Just curious.   

Quote from: gez on March 15, 2007, 07:21:21 PM

More important (IMO) is the following:

http://en.wikipedia.org/wiki/Transition_time

All helps though.

Thanks!  :)
I compared the data sheets for the 4070 and the 4093 and it seems that transition times are identical between the two. Nevertheless, I will try to free some space on my breadboard to compare their performance.

Quote from: gez on March 15, 2007, 07:51:20 PM
It's interesting that you have both inputs of the LM311 biased to Vref.  I do something similar quite often using CMOS output amps and in their quiescent state, the outputs actually try to bias to half supply - though they're far from stable and swing all over the place - and the chips have a tendency to act as an amplifiers towards the tail end of notes, rather than comparators.  Because there's no negative feedback, the signal is always clipped, but the sides lean inwards as a signal decays.  This gives the effect of introducing dynamics - not in terms of amplitude but 'timbral' dynamics.  The end result makes for a more interesting comparator fuzz, but the signal needs cleaning up to be suitable for, say, driving the clock input of a flip-flop, which is why I use Schmidtt trigger buffers.

Do you own a scope?  I'd love to know if something similar happens with these chips (just play a note and watch the slope of the square output as the note decays).  Just curious.   

I only have a software scope I use with my USB audio interface but I will check. Do you think that wiring the LM311 this way might be not optimal for squaring (the squared signal is fed into the PLL only for about 100 ms following each pluck)?

Markus

Quote from: markusw on March 16, 2007, 03:11:42 AM

Quote from: gez on March 15, 2007, 07:21:21 PM

More important (IMO) is the following:

http://en.wikipedia.org/wiki/Transition_time

All helps though.

Thanks!  :)
I compared the data sheets for the 4070 and the 4093 and it seems that transition times are identical between the two. Nevertheless, I will try to free some space on my breadboard to compare their performance.

Quote from: gez on March 15, 2007, 07:51:20 PM
It's interesting that you have both inputs of the LM311 biased to Vref.  I do something similar quite often using CMOS output amps and in their quiescent state, the outputs actually try to bias to half supply - though they're far from stable and swing all over the place - and the chips have a tendency to act as an amplifiers towards the tail end of notes, rather than comparators.  Because there's no negative feedback, the signal is always clipped, but the sides lean inwards as a signal decays.  This gives the effect of introducing dynamics - not in terms of amplitude but 'timbral' dynamics.  The end result makes for a more interesting comparator fuzz, but the signal needs cleaning up to be suitable for, say, driving the clock input of a flip-flop, which is why I use Schmidtt trigger buffers.

Do you own a scope?  I'd love to know if something similar happens with these chips (just play a note and watch the slope of the square output as the note decays).  Just curious.   

I only have a software scope I use with my USB audio interface but I will check. Do you think that wiring the LM311 this way might be not optimal for squaring (the squared signal is fed into the PLL only for about 100 ms following each pluck)?

Markus

Only way to tell is by scoping it to see...or make a comparison and use you ears. 

Same with the 4070 vs 4093 (data sheets schmata sheets)

QuoteOnly way to tell is by scoping it to see...or make a comparison and use you ears. 

Same with the 4070 vs 4093 (data sheets schmata sheets)

Will check it out..... :)

Markus

Just did a few corrections and added the LED indicators in the schem. The file in the link above is updated.
Some more comments:
Values for R3, R4, R21, R14 and R15 are estimations of trim pot settings
R21 is a 100k plus a 10k in series. A multi-turn pot would probably be a good idea. It seems that the minimum sampling time required to achieve reliable lock-in with each pluck can be tweaked for a couple of ms.

Suggestions for improvments are highly appreciated.

The comparator around U6 I suppose could be skipped by just swapping the connections to the inputs of U7.

Markus

Here's another short sample. This time it's just the PLL two octaves up w/o the Ring Mod and on the other channel (as suggested by toneman  :) ) the clean guitar signal.
http://www.aronnelson.com/DIYFiles/up/PLL_SH_2_octaves_up_2_18-03-07.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_SH_2_octaves_up_2_18-03-07.mp3)

Do you think that the glitch at the beginning of each note is tolerable?

Edit: Again I didn't use a pick because I never use it. For any reason however, lock-in is faster with a pick. Don't know why. Probably, the squared signal is somehow "purer" immediately after a pluck if a pick is used and therefor the PLL locks in faster?. Any ideas??

Regards,

Markus

Cool stuff, I've been looking into pluck detectors recently to use with a ADSR controlled filter experiment.
Yours looks interesting, probably overkill for what I need though.
The sample sounds cool, the glitch at the start of the notes is fine by me  :)

Markus,

Quote
Do you think that the glitch at the beginning of each note is tolerable?

I presume you are planning to feed this into some kind of synthy gizmo with an ADSR or at least A-R controlling a filter and envelope?

Lots of real world sounds have a glitch on the attack that gives it character. Synthesists used to put a little short burst of noise or other quasi-harmonically-related something on the start of Sax patches to simulate the way a real saxaphone attack sounds. Fed through the filter, it sounded pretty convincing - at least for the days before samplers came along. I think your glitch will do admirably. Could sound like a micromoog. :icon_biggrin:

Your pluck detector can do double duty triggering the ADSR generator for the synth modules. Got any old SSM or CEM chips laying around? Or you can do discrete if you don't mind all the components. Could be a really cool and unique gizmo.

The VCO tracking is much better than I had anticipated - especially with the bass input. Craig Anderton would be proud. Move over AMS-100. New kid's in town :icon_mrgreen:

Keep up the good work and keep us posted.

Quote from: slacker on March 18, 2007, 09:48:30 AM
Yours looks interesting, probably overkill for what I need though.

I needed a pluck detector that works as reliable as possible. To me it also looks a bit like overkill, considering it's complexity compared to the PLL per se. I just didn't find an easier way with the same performance..

QuoteCool stuff, I've been looking into pluck detectors recently to use with a ADSR controlled filter experiment.

Quote from: MR COFFEE on March 18, 2007, 11:34:31 AM
I presume you are planning to feed this into some kind of synthy gizmo with an ADSR or at least A-R controlling a filter and envelope?

Adding an ADSR (in combination with a VCA to add some envelope to the PLL signal) will be the next project.  ;) I'm not sure whether I'll also add an envelope filter because basically I could use my Meatball clone with it's send/return loop. On the other hand, having it all in one box (rack enclosure?) would be cool too.

For the moment I'm still planning to feed the PLL signal as a carrier into a 2-xfmr/4 -diode ring modulator, actually kind of a "harmonic ring modulator".
Next steps will be to add the divider circuits from the EMM synth on breadboard. It will just be slightly modified to divide by twice the values of the EMM synth. This gives me a factor 16 over the 3rd octave up to allow the use of an shift register for sine generation. Probably square wave is better for the ring modulator anyway but I just wan't to check out whether it might be an option to add.

QuoteLots of real world sounds have a glitch on the attack that gives it character. Synthesists used to put a little short burst of noise or other quasi-harmonically-related something on the start of Sax patches to simulate the way a real saxaphone attack sounds. Fed through the filter, it sounded pretty convincing - at least for the days before samplers came along. I think your glitch will do admirably. Could sound like a micromoog.

Your pluck detector can do double duty triggering the ADSR generator for the synth modules. Got any old SSM or CEM chips laying around? Or you can do discrete if you don't mind all the components. Could be a really cool and unique gizmo

Thanks a lot for your explanations!! Makes we want to get started with planning the next project  ;)

QuoteThe VCO tracking is much better than I had anticipated - especially with the bass input.

Sorry to dissapoint you.  :icon_redface:
This was just a cheap Strat copy, maybe with highs turned down. For bass the pluck detector as well as the PLL would need some cap adjustments. I think Craig Anderton did quite some tweaking to get the thing working with this performance. Probably just C4 (referring to the AMS-100 schem in Device) needs some tweaking to reduce ripple with bass a bit. What do you think?
I will give it a try with bass but I fear the glitch will still be very noticeable. At least too much to sound cool with the ring modulator.

QuoteThe sample sounds cool, the glitch at the start of the notes is fine by me

QuoteKeep up the good work and keep us posted.

Thanks for your motivation!  :)

Markus

I just gave it a try with bass. Basically, increasing C4 to 470n in the AMS-100 schem was sufficient to give a good performance for the  pluck detector.
Besides that I just had to increase the VCO timing cap and unfortunately the sampling time. So for faster runs the PLL doesn't lock in quick enough. For slower runs it works quite OK but the glitch is more noticeable than with guitar.  For some slow attack synth sounds I think it should work fine with bass too.

BTW, I tried to measure the sample time necessary with guitar for reproduceable lock-in by feeding the NE555 pluse into my sound card scope. It was around 120 ms. 
So playing 16th's at 120 bpm should be possible.  ;)

Markus

Quote from: markusw on March 18, 2007, 12:36:27 PM
For the moment I'm still planning to feed the PLL signal as a carrier into a 2-xfmr/4 -diode ring modulator, actually kind of a "harmonic ring modulator".

Hats off to you, this was something I've wanted to do for a long time.  A tracking ring modulator.  Very smart, because it makes the ring modulator useful for more than weird robot sounds (no offense, robots).  How about a footswitch for an option that makes the sample and hold simply hold its note indefinitely, so then you could have it sample a note, get the frequency, but then switch it to hold that note and play other notes in that scale for the different types of nutso (but still harmonic) harmonies with the ring modulator?

Not much to aid your valiant efforts, but its exciting to hear about, and your PLL synth seems to work surprisingly well.  You're in a position to throw in a killer tracking octave down if you wanted, since you've done the hard part already of extracting the fundamental (and even if it isn't the fundamental it would still sound more natural than something jumping around).  Though a combination of settings on the ring mod could probably deliver something along those lines, I like octave downs for the ring modiness anyway, not because bass players haven't been invented yet.

PS I like the glitch.  I haven't messed with PLL's yet but with the basics I'm guessing something could be worked out by adding a second PLL and just using the VCO, and drive that from the first PLL, but lowpass filter the CV so that the glitching and sharp transition between notes is a glide, kind of a portamento sound. 

Also here was an interesting post (http://www.diystompboxes.com/smfforum/index.php?topic=55416.msg426486#msg426486) that came up today that could probably be of interest if you wanted to use your PLL but wanted a sine wave carrier.

QuoteHow about a footswitch for an option that makes the sample and hold simply hold its note indefinitely, so then you could have it sample a note, get the frequency, but then switch it to hold that note and play other notes in that scale for the different types of nutso (but still harmonic) harmonies with the ring modulator?

I planned to include a "hold" stomp switch.  ;) The LF398 with the 1µ cap and the 2000 ohm resistor should hold the frequency for some time. Reminds me that I want to check the droop rate of the LF398....

QuoteYou're in a position to throw in a killer tracking octave down if you wanted, since you've done the hard part already of extracting the fundamental (and even if it isn't the fundamental it would still sound more natural than something jumping around).

It's also planned to be included in the final version  ;) Basically, selecting the carrier will be similar to the EMM synth: -2, -1, 0, +1, +2 +3 octaves as well as 3rd and 5th.
Feeding a flip-flop-divider octave down as a carrier into the ring modulator already sounds pretty cool .

QuotePS I like the glitch.  I haven't messed with PLL's yet but with the basics I'm guessing something could be worked out by adding a second PLL and just using the VCO, and drive that from the first PLL, but lowpass filter the CV so that the glitching and sharp transition between notes is a glide, kind of a portamento sound.

Cool idea!  8) I think we're heading towards a rack unit.....or at least a pretty large box ;) Actually one opamp buffer followed by a low pass filter should be sufficient to drive the second PLL. Will just require some fine tuning to adjust the frequency of the second PLL.  What do you think?

QuoteAlso here was an interesting post that came up today that could probably be of interest if you wanted to use your PLL but wanted a sine wave carrier.

Quote from: markusw on March 18, 2007, 12:36:27 PM

Next steps will be to add the divider circuits from the EMM synth on breadboard. It will just be slightly modified to divide by twice the values of the EMM synth. This gives me a factor 16 over the 3rd octave up to allow the use of an shift register for sine generation. Probably square wave is better for the ring modulator anyway but I just wan't to check out whether it might be an option to add.

;)

Markus

Quote from: Processaurus on March 19, 2007, 05:42:10 AM
  How about a footswitch for an option that makes the sample and hold simply hold its note indefinitely, so then you could have it sample a note, get the frequency, but then switch it to hold that note and play other notes in that scale for the different types of nutso (but still harmonic) harmonies with the ring modulator?

One of the problems I had when I did that was that the note tends to detune over time when it's held for any length of time.  Even using low-leakage switches.  Not significantly, but enough to cause tuning problems.  Also, with stringed instruments it's easy to play the note slightly sharp as you 'capture' it (slight bend/bit of vibrato without thinking) and that's what you're stuck with!  Not to deter anyone from doing this, just that the approach has limitations.

Quote from: gez on March 19, 2007, 09:45:25 AM
One of the problems I had when I did that was that the note tends to detune over time when it's held for any length of time.  Even using low-leakage switches.  Not significantly, but enough to cause tuning problems.  Also, with stringed instruments it's easy to play the note slightly sharp as you 'capture' it (slight bend/bit of vibrato without thinking) and that's what you're stuck with!  Not to deter anyone from doing this, just that the approach has limitations.

Hey Gez,

thanks a lot for sharing your observations!!  :)
In order to reduce the droop rate I'm using a LF398 with a 1 µ hold cap and an additional 2k resistor between the Chold pin and the hold cap. If my calculations are correct, droop rate should be in the range of 5 µV/sec. Probably still too much. Will check this evening with a frequency counter.
What kind of sample/hold circuit did you use in your experiments?

Regards,

Markus

Way out of my thinking and knowledge but I just wanted to drop a quick note to say that I like what I've heard so far.
The ability to trck the pitch with a rimg mod makes it very usable.
Thanks for all the work on this. It's looking...er..sounding nice!

John

Quote from: markusw on March 19, 2007, 10:22:43 AM

Quote from: gez on March 19, 2007, 09:45:25 AM
One of the problems I had when I did that was that the note tends to detune over time when it's held for any length of time.  Even using low-leakage switches.  Not significantly, but enough to cause tuning problems.  Also, with stringed instruments it's easy to play the note slightly sharp as you 'capture' it (slight bend/bit of vibrato without thinking) and that's what you're stuck with!  Not to deter anyone from doing this, just that the approach has limitations.

Hey Gez,

thanks a lot for sharing your observations!!  :)
In order to reduce the droop rate I'm using a LF398 with a 1 µ hold cap and an additional 2k resistor between the Chold pin and the hold cap. If my calculations are correct, droop rate should be in the range of 5 µV/sec. Probably still too much. Will check this evening with a frequency counter.
What kind of sample/hold circuit did you use in your experiments?

Regards,

Markus

OK, I've finally got round to looking at your circuit in detail and refreshing my memory as to what it is you're trying to do (and actually doing by the sound of things).  Because your schematic is a little disjointed it's not easy to follow (for me at any rate), but from my understing based on your notes, the PLL is already sustaining indefinitely and will only change frequency when another 'note event' occurs.  Yes?  If that's the case, all you have to do is disable the 'pluck detector' (break the signal to it with your switch) so that it can't change anything.

You're already using a 4066 for the sample and hold, which has 'infinite' impedance when closed, so providing you don't use an electrolytic for the 1u cap (my preference would be for a larger resistor and smaller cap as they tend to be lower leakage) it should keep pitch for a reasonable amount of time.

i've built the ams 100 in perfboard, the whole project, and the triggered was too slow. Can you tell me what did you change to get fast response?. The filter is really good, and the vca/ring too. Has problems with the lfo square wave though (ticks, assymetry).

Quote from: gez on March 19, 2007, 01:22:23 PM

OK, I've finally got round to looking at your circuit in detail and refreshing my memory as to what it is you're trying to do (and actually doing by the sound of things).  Because your schematic is a little disjointed it's not easy to follow (for me at any rate), but from my understing based on your notes, the PLL is already sustaining indefinitely and will only change frequency when another 'note event' occurs.  Yes?  If that's the case, all you have to do is disable the 'pluck detector' (break the signal to it with your switch) so that it can't change anything.

You're already using a 4066 for the sample and hold, which has 'infinite' impedance when closed, so providing you don't use an electrolytic for the 1u cap (my preference would be for a larger resistor and smaller cap as they tend to be lower leakage) it should keep pitch for a reasonable amount of time.

Hey Gez,

thanks for your help!  :)
Regarding the hold switch I thought of connecting CV1 to gnd for the manual hold (maybe with a 1k resistor between the output of the NE555 and CV1). Bad idea??

I also tried to just use the 4046 loop filter's cap for the sample/hold but voltage was dropping too fast. Probably this ist just a breadboard artifact, don't know. The LF398 keeps the voltage better but still droop rate is higher than calculated. Maybe it will improve on PCB? Any ideas?

Quote from: swt on March 19, 2007, 01:57:52 PM
i've built the ams 100 in perfboard, the whole project, and the triggered was too slow. Can you tell me what did you change to get fast response?. The filter is really good, and the vca/ring too. Has problems with the lfo square wave though (ticks, assymetry).

Hey swt,

basically I breadboarded the AMS-100 pluck detector as in the schem. It was just adapted to a single 12V supply.
Referring to the original schem in "Device":

* R2, R11 and R14 were replaced by a 20k pot with pins 1 and 3 connected to V+ and gnd
* The 20k pot was then adjusted to give Vref at pin 13 of IC1B
* Pins 9 of IC1B and IC3C were connected to Vref
* R12 is 8.2k and R27 is 20k (I didn't have any 10k trim pot left)
* R40 was connected to Vref instead of gnd
* R18 was connected to Vref instead of gnd
* D5 was connected to Vref instead of gnd
* R10 was connected to Vref instead of gnd
* There is a bug in the schem: IC4A pins 1 and 2 should be labelled the other way round
*Vref is done with a 2x 10k voltage divider and an opamp buffer


The output of IC4B was fed into another comparator (see my schem). R14 and R15 (in my schem) actually is a 20k pot. The pot was set to bias the non-inverting input of U6 slightly higher than the output of the opamp in front of it (without a signal).


Otherwise, I followed Craig Anderton's advices on using accurate components for the critical parts and on adjustment of the thing.

Hope this helps  :)

Regards,

Markus

Quote from: markusw on March 20, 2007, 03:40:29 AM
Regarding the hold switch I thought of connecting CV1 to gnd for the manual hold (maybe with a 1k resistor between the output of the NE555 and CV1). Bad idea??

A 1k resistor on the output of the 555 is going to waste a lot of current.

If it were me, I'd connect a resistor (100k shouldn't present a problem - famous last words!) between the output of 555 to input of 1st inverter and take that input as being the C1 ref point.  Then, if you connect a smaller resistor from that point to ground with a momentary , it'll form a divider with the larger resistor and pull the switch's control gate down.

Even better (the low current option and less delay) would be to use two gate logic instead of those inverters, so that the gate functions as an inverter with one input high, but when pulled to ground with a momentary switch the device's output switches everything as you want it.  Too early in the morning to tell you what to use, so I recommend a scan thru Don Lancaster...

   

Quote from: gez on March 20, 2007, 04:46:48 AM

A 1k resistor on the output of the 555 is going to waste a lot of current.

If it were me, I'd connect a resistor (100k shouldn't present a problem - famous last words!) between the output of 555 to input of 1st inverter and take that input as being the C1 ref point.  Then, if you connect a smaller resistor from that point to ground with a momentary , it'll form a divider with the larger resistor and pull the switch's control gate down.

Even better (the low current option and less delay) would be to use two gate logic instead of those inverters, so that the gate functions as an inverter with one input high, but when pulled to ground with a momentary switch the device's output switches everything as you want it.  Too early in the morning to tell you what to use, so I recommend a scan thru Don Lancaster...

   

Thanks a lot!
Didn't realise that a 1k would draw too much current. The "two gate logic" variant sounds far more elegant.
Will have a look at Lancaster.....

Markus

Quote from: markusw on March 20, 2007, 05:03:51 AM

Didn't realise that a 1k would draw too much current. The "two gate logic" variant sounds far more elegant.

The draw won't be constant, you'll get 9mA spikes (assuming a 9V supply: can't remember what you're using) everytime a note event occurs for as long as the switch is depressed.

You shouldn't need anything more that a NOR or NAND gate (can't remember which would be best).

Marcus, I'm probably missing something, but I'm wondering why you have the 555 there at all.

If you look at the section on 'half monostables' in Lancaster's CMOS Cookbook you'll see that C6 and R23 are enough to do the job of providing a pulse of the duration you require (though you'll probably need to tweak values).

If you use those components in conjunction with a dual-input logic gate you'll be able to ditch the 555 and have the ability to disable the pulses to the switches so that the sample and hold is always 'locked'.  Plus you'll be able to debounce the switch easily.

PS  Whenever you have an arrangement such as C6 and R23 it's always best to wire a resistor between the junction of those two components and the input of the logic device you're using, to protect said device.  Although you obviously haven't damaged your 555 as it's working, I have damaged a few logic chips in my time by being lazy/forgetful and not using this protection.

Quote from: gez on March 20, 2007, 04:46:48 AM
Even better (the low current option and less delay) would be to use two gate logic instead of those inverters,

That should have read 'two input logic'

Quote from: gez on March 20, 2007, 05:39:51 AM
Marcus, I'm probably missing something, but I'm wondering why you have the 555 there at all.

If you look at the section on 'half monostables' in Lancaster's CMOS Cookbook you'll see that C6 and R23 are enough to do the job of providing a pulse of the duration you require (though you'll probably need to tweak values).

If you use those components in conjunction with dual-gate logic you'll be able to ditch the 555 and have the ability to disable the pulses to the switches so that the sample and hold is always 'locked'.  Plus you'll be able to debounce the switch easily.

PS  Whenever you have an arrangement such as C6 and R23 it's always best to wire a resistor between the junction of those two components and the input of the logic device you're using, to protect said device.  Although you obviously haven't damaged your 555 as it's working, I have damaged a few logic chips in my time by being lazy/forgetful and not using this protection.

Thanks a lot!

Didn't think about this way of pulse stretching. I used the 555 because it provides an easy way of adjusting the sample time. So if one would just use C6 and R23 for determining the sample time one would simply replace R23 with a pot so that sampling time can be tweaked??

Thanks also for warning me about the missing protection resistor at the input of the 555. Think I should read the Lancaster book  ;)

BTW, from the 4070 to the input of the 4046 there is also a resistor I forgot to add to schem. IIRC it's a 10k. Without the resistor the PLL doesn't work that great. Any ideas why?

QuoteThat should have read 'two input logic'

OK, thanks!  :)

Markus

Quote from: markusw on March 20, 2007, 06:12:46 AM
So if one would just use C6 and R23 for determining the sample time one would simply replace R23 with a pot so that sampling time can be tweaked??

Sure, or you could calculate the value needed.  Whatever is easiest.

QuoteBTW, from the 4070 to the input of the 4046 there is also a resistor I forgot to add to schem. IIRC it's a 10k. Without the resistor the PLL doesn't work that great. Any ideas why?

I'd have to look at the 4046 schematic (and your schematic) in detail.  What value is the resistor?  Off the top of my head though, no I can't think why that would be.

If you've got a 4093 (think you said you had one) it could replace the 4070 and your two inverters, plus you've now got your 2 input logic gate for disabling the pluses to the sample and hold circuitry.

Or if you haven't got a 4093 you could use a 40106 and some diodes and a resistor to make a 2 input gate. That would also give you 6 inverters in total, which might be able to replace other things as well.
You probably already know all about diode logic but if not have a look here.
http://www.musicfromouterspace.com/analogsynth/mmlogic.html (http://www.musicfromouterspace.com/analogsynth/mmlogic.html)

QuoteIf you've got a 4093 (think you said you had one) it could replace the 4070 and your two inverters, plus you've now got your 2 input logic gate for disabling the pluses to the sample and hold circuitry.

Thanks!  :) Sounds great. Will check it out for sure. Would be cool to reduce the number of chips!

QuoteOr if you haven't got a 4093 you could use a 40106 and some diodes and a resistor to make a 2 input gate. That would also give you 6 inverters in total, which might be able to replace other things as well.
You probably already know all about diode logic but if not have a look here.
http://www.musicfromouterspace.com/analogsynth/mmlogic.html

I have a 4093. Nevertheless, thanks!
"Know" about mm logic is probably not correct, "heard of", yes  ;) Will check it out more thoroughly now... Thanks for the link!

Markus

Just had a brief look at the 4093 data sheet. It includes the calcs for exactly what you suggested. A 1µ cap plus a 100k to 1 Meg pot should be fine to tweak the sample time.
Also it would be retriggerable which I prefer anyway. So incase you would play faster than the "sample time" it would extend the sample time (if I understood it correctly  ;) ).

Markus

Quote from: markusw on March 20, 2007, 10:21:06 AM
Just had a brief look at the 4093 data sheet. It includes the calcs for exactly what you suggested. A 1µ cap plus a 100k to 1 Meg pot should be fine to tweak the sample time.
Also it would be retriggerable which I prefer anyway. So incase you would play faster than the "sample time" it would extend the sample time (if I understood it correctly  ;) ).

Markus

Sure you're not confusing this chip with the 4098 (retriggerable monostable)?  Or does the data sheet for the 4093 (which is a 2 input NAND with schmidtt inputs) show some applications?

Incidentally, you could probably use the 4070 to do what I suggested (replace the inverts & disable pulses to S&H), just that I think the 4093 is a better chip for this purpose.

Quote from: gez on March 20, 2007, 11:22:17 AM
Sure you're not confusing this chip with the 4098 (retriggerable monostable)?  Or does the data sheet for the 4093 (which is a 2 input NAND with schmidtt inputs) show some applications?

Incidentally, you could probably use the 4070 to do what I suggested (replace the inverts & disable pulses to S&H), just that I think the 4093 is a better chip for this purpose.

Now I'm sure I'm confusing something.  ;) Obviously it doesn't work the way I want. Don't know if the correct expression is retriggerable.
The idea is: let's say the sample time is set to 100 ms. Now let's assume you play two notes with just 90 ms between them. With the NE555 wired like it is at the moment the second note wouldn't trigger. If I wire the NE555 like in the Electrax synth (as a attack/decay generator) it would be retriggered, thus the sampling time would be extended to 190 ms.
Is there a way to make it "re-triggerable" with a 4093?

Thanks again for your help!!

Markus

OK Markus, understood.

You might be able to wire up a 4093 as a retriggerable monostable (with a few additional components) but it won't be pretty.

The 555 could easily be made retriggerable, but it would require DC coupling.  No big deal as R23 & C6 seem superfluous.  However, I'd use a 4098.  It would replace the 555 and the inverters as it has outputs that oppose one another (when one output is logic 1 the other is logic 0).  You can also use the reset pin to disable pulses (which you can also do with the 555 incidentally - don't know why that didn't occur to me earlier).

Only draw back with the 4098 is that its outputs can't source/sink much current so you'd need to use a discrete FET/trannie for the LED.

It might be possible to wire the other device in the 4098 as an inverter/buffer to do the job of the 4070, but I doubt it would perform that well...who knows?

http://hammer.ampage.org/files/EH-Guitar-Synth.ZIP
See board 1 in the zip file for an example of 4098 use.

QuoteThe 555 could easily be made retriggerable

How would you do this?  :icon_redface:
The only way I found so far was the attack/decay generator of the Electrax synth but it requires a comparator to provide low and high signals, at least a comparator was the only thing that came to my mind  ;)

Quoteas it has outputs that oppose one another (when one output is logic 1 the other is logic 0).

Definitely, an advantage over the 555.

QuoteYou can also use the reset pin to disable pulses (which you can also do with the 555 incidentally - don't know why that didn't occur to me earlier).

Thanks for the tip!  :)

QuoteOnly draw back with the 4098 is that its outputs can't source/sink much current so you'd need to use a discrete FET/trannie for the LED.

This would still be fine since a trannie plus a resistor or two doesn't take much space.

Quote from: StephenGiles on March 20, 2007, 05:45:02 PM
http://hammer.ampage.org/files/EH-Guitar-Synth.ZIP
See board 1 in the zip file for an example of 4098 use.

Thanks!! Will have a look at it.  :)

Markus

The 555 is negative edge triggered, so you need to find a way to discharge the cap everytime the input goes low.  If DC coupled, the easiest way to do this is with a diode.  Connect the cathode to the input (pin 2) and the anode to the junction of the cap and resistor (pin 7).  This is the cheapskate way of doing things and it has three drawbacks:

1. The cap only discharges to 0.6V so timing is affected (doubt its critical here) and you have to compensate by making the resistor slightly larger in value, and

2.  You have to watch that the device preceding the 555 doesn't sink current to the point where it's damaged.  With a 30k resistor, you shouldn't have a problem (but always best to check data sheets).

3.  You'd need to fix the preceding comparator to be 'high' when in its quiescent state

The better way to do this (the method you find in text books) is using a PNP trannie.  Connect the base to the input, its emitter to pin7 and its collector to ground.  A p-channel MOSFET could also be used.  This is the best option for you as you won't have to bias the comparator to be high - keep R23 and C6.

Thanks a lot, Gez!!!  :) :)

Markus

PS  Stick a Schottky diode in parallel with R23 - cathode to V+, anode to junction of C6, R23 and pin 2 - to give protection to the trannie & the 555 (shouldn't be any need for the additional protective resistor with the diode wired in there).

Quote from: gez on March 21, 2007, 09:26:51 AM
PS  Stick a Schottky diode in parallel with R23 - cathode to V+, anode to junction of C6, R23 and pin 2 - to give protection to the trannie & the 555 (shouldn't be any need for the additional protective resistor with the diode wired in there).

Thanks again!  :)
Could I also use a 1N4184 instead of the Schottky?

Quote3.  You'd need to fix the preceding comparator to be 'high' when in its quiescent state

I'm confused now. The precceding comparator is high when in quiescent state.

Markus

Quote from: markusw on March 21, 2007, 10:08:01 AM

Quote3.  You'd need to fix the preceding comparator to be 'high' when in its quiescent state

I'm confused now. The precceding comparator is high when in quiescent state.

My apologies, I didn't look at your schematic in detail.  If that's the case, you can simply use a diode - pull C6 and R23 and DC couple the 555 (no need for protection diodes/resistors) - or the trannie if you want more precision (again, pull all the other stuff - it's superfluous).

Quote from: gez on March 21, 2007, 10:12:33 AM
My apologies, I didn't look at your schematic in detail.  If that's the case, you can simply use a diode - pull C6 and R23 and DC couple the 555 (no need for protection diodes/resistors) - or the trannie if you want more precision (again, pull all the other stuff - it's superfluous).

Thanks for the clarification!  :)

BTW, I found a schem on how to convert a 555 to retriggerable. Unfortunately, it's in german but there are schems: http://www.elektronik-kompendium.de/public/schaerer/retr555.htm (http://www.elektronik-kompendium.de/public/schaerer/retr555.htm)

It seems to do exactly what I want  but it is far more complex than you suggested.
Will check out your suggestions for sure because they are much simpler. LTSpice will help me understanding what's going on  ;)

Markus

What do you think about this possibility? It's the attack/decay generator from the Electrax plus some 4093's....

(http://img.photobucket.com/albums/v21/markusw/NE555retriggerableplus4093.jpg)

It would be retriggerable and I could use one of the 4093's for cleaning up the squared guitar signal.

Regards,

Markus

Quote from: markusw on March 22, 2007, 02:43:01 AM
It would be retriggerable

Maybe I'm missing something, but I don't see how it is.  :icon_confused:

As I've already mentioned, if you've biased the preceding comparator so that it's high when no signal is present, you don't need C6 and R23.

Quote from: gez on March 22, 2007, 05:08:38 AM
Maybe I'm missing something, but I don't see how it is.  :icon_confused:

As I've already mentioned, if you've biased the preceding comparator so that it's high when no signal is present, you don't need C6 and R23.

It's taken from the Electrax schem page 17 http://experimentalistsanonymous.com/diy/Uploads/electrax.pdf (http://experimentalistsanonymous.com/diy/Uploads/electrax.pdf). The description how the NE555 stage works is also in the file. It's for sure better than I could explain  ;)

At least according to the LTSpice sims it should be retriggerable. C6 and R23 are also derived from the Electrax. If I understand it correctly C6 and R23 just make very short pulses from the comparator's output. So in comparison to feeding the precceding comparator directly into the 555 the pulse generated by the 555 should be more precise (time-wise). E.g. depending on picking strength the pulse length from the 555 would vary when fed directly with the comparator's out (e.g. let's assume the pulse generated from the comparator are 100 ms and the 555 is set to give a 100 ms pulse it would be 200 ms in total). Maybe it's just rubish  :icon_redface:

Markus

Quote from: markusw on March 22, 2007, 08:22:36 AMThe description how the NE555 stage works is also in the file.

What page?  I'm afraid I don't have the time to wade thru all that!

QuoteIf I understand it correctly C6 and R23 just make very short pulses from the comparator's output. So in comparison to feeding the precceding comparator directly into the 555 the pulse generated by the 555 should be more precise (time-wise). E.g. depending on picking strength the pulse length from the 555 would vary when fed directly with the comparator's out (e.g. let's assume the pulse generated from the comparator are 100 ms and the 555 is set to give a 100 ms pulse it would be 200 ms in total).

OK, from what I gather, you're trying to make it as close to positive edge-triggered as possible by reducing the pulses to a slither, so that triggering (which happens on the negative going edge) occurs quickly.  Still seems unnecessary to me.  I don't have time to go thru you circuit in detail, but if it's vital that triggering happens in phase with another side chain then you could probably just invert the phase of a preceding stage to the 555.  Maybe I'm missing something.

QuoteWhat page?  I'm afraid I don't have the time to wade thru all that!

It's on page 6 under "Envelope generator". BTW, the pages numbers are the total page numbers given by Adobe Reader not the ones printed on the pages.

QuoteOK, from what I gather, you're trying to make it as close to positive edge-triggered as possible by reducing the pulses to a slither, so that triggering (which happens on the negative going edge) occurs quickly.  Still seems unnecessary to me.  I don't have time to go thru you circuit in detail, but if it's vital that triggering happens in phase with another side chain then you could probably just invert the phase of a preceding stage to the 555.  Maybe I'm missing something.

Basically, I just want the pulses to have the same length irrespective of the pulse lenght generated by the preceeding comparator.
Sorry, why would inverting the phase of the comparator help?  :icon_redface: It goes negative with each pluck. So I could leave C6 and R23 off but then sampling time would depend on picking strenght (at least to some extend).
BTW, I already added a protection diode from the 555's input to V+.

Markus
 

hey M,

haven't seen that Electrax article in a LONG time!!   :P
thanx for reminding me    :D
good article!!!   definitely a "modular guitar"    :icon_biggrin:
and....there's your VCA!!!   U could sub a dual OTA like the 13700/13600 easily.

but, reguarding:
missing pulse detectors, ADSRs, ADs, retriggerable one-shots, LED pulse stretchers, threshold comparators and VCOs--
the 555 will do it all   ;)
There are CMOS versions as well.
afn
T

I've also built the electrax in perfboard. It's triggered is slow as a turtle. But the attack realease function works well. It would be nice to get it working with the ams trigger. By the way markus...can you tell me what can be wrong in the lfo of the electrax?. it's square wave it's somewhat strange. it also ticks like hell. Thanks.
By the way, the harmonic generator also works fine, and the octaves down tracks better than some pedals, but have to do some mods to get it right. Filter is amazing, and also vca with ring mod type sounds, although carrier bleeds a little, specially with squarewaves.

Quote from: markusw on March 22, 2007, 08:58:08 AM
Basically, I just want the pulses to have the same length irrespective of the pulse lenght generated by the preceeding comparator.
Sorry, why would inverting the phase of the comparator help?  :icon_redface: It goes negative with each pluck. So I could leave C6 and R23 off but then sampling time would depend on picking strenght (at least to some extend).

How would it depend on picking strength?  A comparator only has two states: high and low.

I don't think I fully understand what it is you're trying to do.  Why do you need all the pulses to have the same time Length from the initial attack of a note to the finish of the one-shot's timed period?  If you're going to make the 555 retriggerable its output will always be high, and the circuit always sampling, until the signal drops below a threshold set by the comparator (I'm presuming you've done this); only then will the 'hold' part happen.  I somehow get the feeling this isn't what you want to achieve?

QuoteI've also built the electrax in perfboard.

Wow!  :o

QuoteBy the way, the harmonic generator also works fine, and the octaves down tracks better than some pedals, but have to do some mods to get it right. Filter is amazing, and also vca with ring mod type sounds, although carrier bleeds a little, specially with squarewaves.

Good to know! Thanks for the information!  :)

QuoteBy the way markus...can you tell me what can be wrong in the lfo of the electrax?. it's square wave it's somewhat strange. it also ticks like hell. Thanks.

I'm not sure that I'm the right one for asking  ;) but I will have a look at it!

QuoteHow would it depend on picking strength?  A comparator only has two states: high and low.

If picking strenght is higher the comparator would stay high (or low) longer. 

QuoteWhy do you need all the pulses to have the same time Length from the initial attack of a note to the finish of the one-shot's timed period?

I want it to sample just for as much time as necessary to achieve lock-in (should be around 120 ms). So sampling should be done immediately after the initial attack.

QuoteIf you're going to make the 555 retriggerable its output will always be high, and the circuit always sampling, until the signal drops below a threshold set by the comparator (I'm presuming you've done this)

Assuming a sampling time of 120 ms it should only retrigger if one would play faster than one note each 120 ms. If you play slower, it samples for 120 ms and then stays at this frequency for at least some time. The idea is, that due to the four 4066 switches the PC2 out stays at the level of the last note. This way the lock-in  for the next note (provided it's not 2 octaves above or below) will be faster (or at least less noticeable). This works pretty well. If I have the PLL running at idle frequency and play a note the lock-in is more noticeable than if I play the same note a second time (or another close note).

If it would sample as long as the signal is above a certain threshold PC2 out voltage would drop at the end of notes, because the envelope follower, retriggerable monostable or whatever is used for triggering simply isn't fast enough for turning to hold. With a defined sampling time this issue doen't exist.

Hopefully, my explanations are not too confusing.  ;)

Markus

Quoteand....there's your VCA!!!   U could sub a dual OTA like the 13700/13600 easily.

That's for the next module  ;)

I've got an interesting issue at the moment. I replaced the two 4049 inverter stages in the pdf schem with a 4093.
Interestingly, at first it didn't work. OK this isn't really astonishing. After searching for wiring bugs I started checking voltages. Once I connected the DMM to the output of the 555 it started working again, like before I replaced the chip. If I disconnect the DMM it stops working.
I tried to replace the DMM with resistors (10k to 10M) from the NE555's output to either gnd or V+  but it didn't help. Notably, it keeps working as long as the DMM is connected even if I add a 10k to V+ or gnd.
It's really funny because performance is exactly the same as before I replaced the 4049 with a 4093, it just needs the DMM connected to the output of the NE555.

What kind of voodoo is that?  :icon_confused:

Any ideas would be highly appreciated because I need my DMM for something else.  ;)



Markus

Assuming both of the two 4093's inputs are tied together, U have just made a schmidt-trigger inverter. :o
Not really necessary for this application.....but.....
So... U used only TWO sections of the quad NAND gate? to make TWO inverters to replace the TWO 4049 sections??
whew!  almost as long-winded as Mr Hammer   :icon_biggrin:
Did U remember to tie ALL unused CMOS inputs to gnd or pos???   ;)
JUST the inputs..  :icon_exclaim: :icon_exclaim: ......leave unused outputs floating.
lmk
8)
T

QuoteDid U remember to tie ALL unused CMOS inputs to gnd or pos???   
JUST the inputs..    ......leave unused outputs floating.

Yup, the inputs are tied to gnd. Thanks for the tip!  :)

12 h later it still only works if the DMM is connected to the NE555's out?

Any other ideas?

Markus

I found it. It doesn't work if I tap off the CV1 directly from the NE555's out. If I have two 4093 inverters in series connected to the NE555's out  and take CV1 and CV2  from their outputs it works. CV1 is then taken from the second inverter.
Any ideas why it doesn't work to CV1 directly from the 555??

Markus

Here's another update:

http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_schem_24-03-07.pdf (http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_schem_24-03-07.pdf)

Changing the 555 to retriggerable was definitely a good modification. Even if you play faster than sampling time no notes are missed.
I also added a 4015  shift register plus a couple of resistors for sine approximation. With a simpe passive lp filter distortion is around 0.1%.

It's already real fun playing with it.  :)

Next I will try to add the programmable divider from the EMM synth....



swt, I had a look at the Electrax LFO. Maybe decoupling of the LFO opamp power rail might help?? I hope that others with more experience will chime in.



Markus

Very cool Markus!!!

Can't wiat to hear the sound clips. How many steps per cycle in your 4015 sine converter?

Keep us posted.

QuoteHow many steps per cycle in your 4015 sine converter?

Should be 16 steps. It divides by 16.

One Q: I tried to tap off a square wave at 1/16 of the frequency from one of the 4015's outputs but it seems that I can't get a pure square wave from them. I suppose due to the mixing resistors it's a mess. Any ideas how to get the square wave at the same freq as the sine from the 4015 without using a buffer? Also would like to avoid another divide-by-16 chip.....

BTW, the programmable divider from the EMM synth is already working  :) Just have to tweak the 4046 VCO timing cap to adapt to the higher divider.......

QuoteCan't wiat to hear the sound clips.

I will try to da a sample of the sine approximation vs. dry guitar signal  :) First I will have to tweak the PLL to run with a 128 to 192 divider......


Regards,

Markus

none of the outputs of any shift register (4014, 4015 or 4017) have SQwave outs.
it's just the nature of the beast.
To get a 50/50 square wave U need a divide-by-2 each time.
the 4015 is 2 4bit shift registers.
the 4014 is a single 8bit shift register.
Several web sites show the output counts of this shift-registers.

I'd say, use one of the divided clocks to generate a sine using a 4040/R2R technique.
check your email
afn
T

Hey T,

thanks for your help!!

Obviously I had a wiring bug yesterday evening. Now it works. I get a 50/50 square wave (at the same freq as the sine approximation)  from every output of the 4015.
If I take it from the 4th output it's in phase with the sine.

In the meantime I added a 4040 between the 4046's VCO out and the 4015 sine generator to select the various octaves.
At the moment there are about 15 ICs on breadboard. So the PCB(s) won't fit into a 1590BB  :icon_biggrin:

BTW, I intensively checked whether there is any advantage of having a 4093 stage between the 311 guitar signal squarer and the PLL's in, and to be honest, it really seems the 311 is sufficient. At least in my hand I do not get an improved performance if the 4093 is added.
Nevertheless, I probably will leave it in because there is one stage of the 4093 left.

Markus

Could you possibly post a picture of your breadboard as is - I must see this!

Quote from: StephenGiles on March 25, 2007, 06:37:31 AM
Could you possibly post a picture of your breadboard as is - I must see this!

Sure, I will take a pic later. To be honest, looking at this mess it's hard to believe that it works  ;)
So it will for sure be OK for a good laugh  :icon_biggrin:
Due to it's 3D wiring I do not yet have an idea how large the PCB(s) will be.
Markus

Here is a pic of the current mess

(http://img.photobucket.com/albums/v21/markusw/IMG_0825rs-2.jpg)

The most left board contains the AMS pluck detector (the 3 top chips) and an orange squeezer.
The second from the left contains parts of the pluck detector, the NE555, the 4093, the 4016, the LF398 and a LM311 (at the moment used as output buffer).
The 3rd one has the input gain stage, Vref opamps, the guitar signal squarer LM311, the 4046 and the two 4526's for the programmable divider.
The 4th one contains the input lp filter, an additional opamp stage (currently used as output buffer) and mainly some stuff from previous experiments (i.e. an adaptive Schmitt trigger and a 4013 divider) that is not in use at the moment.
The most right board finally contains the 4040 divider and the 4015 sine generator.

Markus

That's amazing  :o
Looks pretty neat and tidy to me.

Amazing indeed, I've been there many times and often something that worked one day would not the next - as happened with the ADA Flanger, so I ripped it all out and started again!

Quote from: slacker on March 25, 2007, 08:22:12 AM
That's amazing  :o
Looks pretty neat and tidy to me.

QuoteLooks pretty neat and tidy to me.

I just hope that it will look a bit more tidy when on PCB.  ;) To be honest, I myself wonder quite often that it works.

BTW, I like tose tiny trim pots. They fit nicely onto two or three rows.

QuoteI've been there many times and often something that worked one day would not the next - as happened with the ADA Flanger, so I ripped it all out and started again!

Sometimes it's really pretty weird, it stops working....you do nothing, maybe unconnect it from the ps....then after a while power it up again and everything's perfect again.
Currently, since the 555 issue was solved it seems pretty stable...no DMM needed to keep it running   :icon_biggrin:

Quoteso I ripped it all out and started again!

I think I would quit the project before wiring up a second time   ;)

Markus

WOW!   :icon_eek:    :o

U gonna go surfacemount????

8)
T

Quote from: toneman on March 25, 2007, 03:05:17 PM
WOW!   :icon_eek:    :o

U gonna go surfacemount????

8)
T

Hopefully not  ;)

Markus

Another update  ;)

Yesterday eve I added a simple gain control on breadboard to add an envelope to the VCO signal. I just wanted to give it a try with a LED/LDR combo  ;)  It worked out astonishingly well.
Basically, I tapped off the signal from the NE571 rectifier (part of the AMS-100 pluck detector) and run it through a non-inverting opamp stage with slight gain. Subsequent to the opamp stage there is a simple R/C low pass consisting of a 20k pot and a 10µ to gnd. Then there is another opamp buffer stage. From it's out there is a trimpot (current limiter) followed by an LED to gnd. The LED is part of a home-made LED/LDR combo. The LDR is wired in series with the output of the sine generator (or the square wave) followed by a 10k to gnd (like a volume pot). It needed just a little tweaking of the gain stage and the current limiting pot to null the signal when in idle state.
Interestingly, the LED/LDR combo seems to smoothen any ripple. So for fast attack the R/C low pass isn't even necessary. With slow attack (pot set to 5-10k) it sounds a bit like a reverse tape effect.
With the sine or square wave set to e.g. 3rd or 5th or suboctave it sounds pretty cool.
Will probably give it a try with a LM13700 VCA to compare it to the LED/LDR combo but actually I would like to stay with the LED/LDR combo....I simply like to use them.

Will try to do some samples this eve. Also the schem urgently needs an update ;-)

Markus

Here are two samples with the LED/LDR gain control. Two are sine, once with one octave down, second one 5th up. Third clip is again 1 octave down, this time square wave.
One channel dry again..

http://www.aronnelson.com/DIYFiles/up/PLL_sine_-1_octave_30-03-07.mp3

http://www.aronnelson.com/DIYFiles/up/PLL_sine_5th_30-03-07.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_sine_5th_30-03-07.mp3)

http://www.aronnelson.com/DIYFiles/up/PLL_square_-1_octave_30-03-07.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_square_-1_octave_30-03-07.mp3)

Regards,

Markus

As per suggestion from toneman (thanks!  :) ) I adapted the levels between left and right channel (wet and dry). Otherwise the samples are the same.
Didn't mention before. It was done with my cheap Strat copy directly into the PC. Also there is no tone control so far....

http://www.aronnelson.com/DIYFiles/up/PLL_sine_-1_octave_30-03-07_3.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_sine_-1_octave_30-03-07_3.mp3)

http://www.aronnelson.com/DIYFiles/up/PLL_sine_5th_30-03-07_4.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_sine_5th_30-03-07_4.mp3)

http://www.aronnelson.com/DIYFiles/up/PLL_square_-1_octave_30-03-07_2.mp3 (http://www.aronnelson.com/DIYFiles/up/PLL_square_-1_octave_30-03-07_2.mp3)

Regards,

Markus

Sounds really cool. Looking forward to the updated schem, not that I'll understand it all.

QuoteLooking forward to the updated schem,

Will try to put it together this weekend...

Quotenot that I'll understand it all.

I think I should draw it a bit less confusing..... ;)

Markus

Here's an updated schem including the LED/LDR gain control, programmable divider and sine generator.
I hope it's not too confusing....

http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch_31-03-07.pdf (http://www.aronnelson.com/DIYFiles/up/PLL_AMS100_PD_NE555_current_sch_31-03-07.pdf)

Regards,

Markus

That's quite a schem you have there...impressive.

Greetings from sunny Mar del Plata, Argentina!! That looks a very interesting circuit.

Quote from: StephenGiles on April 03, 2007, 03:55:45 PM
Greetings from sunny Mar del Plata, Argentina!! That looks a very interesting circuit.

Regards to Argentina  :) Have a nice holiday !

Markus

Here's nother update for the schem:

http://www.aronnelson.com/DIYFiles/up/Harmonic_Ring_Modulator_current_sch_15-04-07.pdf (http://www.aronnelson.com/DIYFiles/up/Harmonic_Ring_Modulator_current_sch_15-04-07.pdf)

Still missing: a two-band tone control. Any suggestions would be highly appreciated  :)

Next steps will be to make sine and sqaure wave blendable so you can have everything from pure sine to square wave as a carrier. Also a distortion stage a la Ring Stinger for the dry signal as well as some blend pots for mixing dry, ring mod, and maybe distorted guitar signal still have to be done......

I was already thinking about which controls might be useful with expression pedal and what other outputs to add as jacks (e.g. the AMS100 pluck detector ouput). Any ideas?

Will try to do some soundsamples for the sine octaves in combination with the RingMod next week.....


Regards,

Markus

Quote from: markusw on April 15, 2007, 01:49:25 PM
Still missing: a two-band tone control. Any suggestions would be highly appreciated  :)

Although I'm utterly intimidated by the complexity of the schem (I'm just an amateur, amateur I tells ya!!!) I vote for Baxandall tone control, popularized by SansAMP, and seen i nthe schematic by Tonepad.com:  http://www.tonepad.com/project.asp?id=22

Hi M,
for a tone control-- i'd go 4 a PAIA 4730
it's a voltage controlled multimode filter..LowPass BandPass & HighPass.
all available at same time.!.  U can combine the HP & LP and make a NotchFilter also   ;)
Sine waves have nothing to filter, but the SQuare waves certainly do.

As for pedal controls---
changing the fuzz changes the timbre of the RingModulation.
Since your carrier, now, is an octave multiple of the fundamental,
using a VoltageControlledFilter (VCF) will allow control via a pedal.   ;)
So,....how about another 4730 here(??)  --after the fuzz...

Since U have most of the circuit(s) for an envelope follower, using an EF to "gate" the RingModulation
or the fuzz, might be interesting....(?)
Kind of like a hybrid of a Meatball-meets-RingO-thru-a-Filter-thingie...

I like the word "thingie" ......verrrrrry "technical".

How big did U say the pcb was going to be????    LOL!!!

afn
8)
Tone

Quotefor a tone control-- i'd go 4 a PAIA 4730
it's a voltage controlled multimode filter..LowPass BandPass & HighPass.

I forgot to say I need a "simple" two band tone control  ;)

QuoteI vote for Baxandall tone control, popularized by SansAMP,

Thanks for your vote  :icon_biggrin: This one gets my vote too :)

The 4730 is too complex for my purpose. I want to finish that thingy soon. So additional modules will have to wait  ;)
Nevertheless, thanks a lot for your help toneman!!

QuoteSine waves have nothing to filter,

The voltage controlled low pass (LM13700) works fine for getting a rather pure sine wave. I just need a "post-ring-mod" tone control... :)

QuoteHow big did U say the pcb was going to be?

Considering that the PCB will be pretty large with the circuit in it's present form I fear I have to skip additional " Meatball-meets-RingO-thru-a-Filter-thingie..." modules  ;)

Regards,

Markus

Impressive project and realization.
I think I'll have to go through all that once or two times more.
Pretty thick indeed.

Sounds really good to me

thank you all

Quote from: ignsk on April 16, 2007, 04:20:16 PM
Impressive project and realization.
I think I'll have to go through all that once or two times more.
Pretty thick indeed.

Sounds really good to me

thank you all

Glad you like it  :)

I think it would help if I made a summary on how the circuit works, or at least I think how it works  ;)

Markus

Quote from: markusw on April 17, 2007, 04:27:13 AM
I think it would help if I made a summary on how the circuit works, or at least I think how it works  ;)

Obviously it would... as far as I'm concerned . I did not went through my second reading yet so I'm still stuck in the maze of those trigs PLL VCO S&H  ??? but I know I 'll make it(Wonderfull hope of the noob  ;) )

can anyone here help me understand this schematic? the lf398 doesnt make much sense to me (the pins are what?) and there is no input processing shown. is there a good low pass filter schematic i could put in front of this project?

any 4046 experts out there? (thats the main section i am interested in is the 4046 and diviiding part

thanks!!!

I'm very interseted in building Marcus W's Strring Pluck Detector. It sounds awsome.  What I'm trying to figure out is the connections to some of the ICs. Can anyone explain what pin numbers are used for the 4015 and the 4040?   

Here's the pinout for the 4040 and 4015:

(https://i0.wp.com/datasheethub.com/wp-content/uploads/2022/01/CD4040-IC-Pinout-Diagram.jpg)

(https://www.build-electronic-circuits.com/wp-content/uploads/2020/11/4015_Pinout-white.png)

Q outputs are easy, Clock inputs are pin 10 on the 4040, 1 and 9 on the 4015.

Be aware that that schematic is from a simulation. So the circuit might be tested in the sim alone, and it doesn't look entirely complete to me - or at least assumes quite a lot of stuff.

Thanks for your quick reply.  What confuses me are the labels Marucs used such as dsa, dsb etc. I/m not familiar with logic circuit labels. Any help you can give explaining this is greatly appreciated 

Quote from: RGP on June 04, 2023, 02:39:16 PM
Thanks for your quick reply.  What confuses me are the labels Marucs used such as dsa, dsb etc. I/m not familiar with logic circuit labels. Any help you can give explaining this is greatly appreciated

It's not a question of being familiar with logic circuits. It's just a question of matching up the labels on the schematic with the pinouts - and not all pinouts will use the same labels. If the schematic had pin numbers, we wouldn't have to bother, but as I said, it's a simulation drawing, not a "proper" schematic from some CAD/CAM software.

For the two you mention, we've got "D_A" and "D_B" on the 4015 pins 7 and 15, so I'd guess those are the ones. I don't know what they do exactly without looking up more about the 4015.

I looked closely at several similar schematics for the E&MM Harmony Generator and I think the 4040 connection labled R might be pin 11 & the one labled CP might be pin 10. On the 4015 CP might be pins 1&9 tied together. MR pins 6&14 tied together. I think you're right about the DSA & DSB being pins 7&15. One other thing. What do the small circles mean?       
I'm redrafting Marcus' schematic in an old graphics program I use. Once I have it done I'll find out how to post it here. Again, thanks much for your help.     

Ah, now, the small circles *are* a proper logic symbol. That is used to indicate an inversion. So for example, a NAND gate symbol (like the 4093) is the same as an AND gate (a curvy box thing) with an extra little circle stuck on the output:

(https://dyclassroom.com/image/topic/logic-gate/nand-nor/nand-gate.png)

In this case, we've got some little circles on some of the *inputs*, indicating that they're inverted logic AKA "Active low". That's pretty common on Chip Enable pins and Reset inputs.

I agree with some of your analysis of the 4015 connections. It definitely looks like the two Clock inputs and the two Reset inputs are tied together. There's quite a bit I don't understand though:

1) Those reset inputs. Stuff I can find online about the 4015 (and that pinout above) shows the Reset inputs as active-high, but this schematic suggests active-low. Tying the resets high like that would stop the chip from doing anything, according to National Semiconductor (https://www.electroschematics.com/wp-content/uploads/2013/08/cd4015-datasheet.pdf).

2) Is this set up as two four-bit shift registers in parallel or one eight-bit shift register?
The two data inputs tied together makes it look like dual 4-bit in parallel, but one 8-bit register makes more sense to me for what it's doing (generating a steppy "sine wave" approximation by outputing a different voltage level each time it's clocked).

Quote from: ElectricDruid on June 05, 2023, 06:11:29 AM
I agree with some of your analysis of the 4015 connections. It definitely looks like the two Clock inputs and the two Reset inputs are tied together. There's quite a bit I don't understand though:

It's a common trick to generate a low distortion sine-waves digitally.

The 4015 is wired as an 8-bit shift register, which you initialize to all zeros or all ones.  Each clock pulse shuffles the 8 bit pattern but the input feeding back into the start of the shift register is inverted.   If you look at any *bit* on the shift register it looks like a square wave with a period of *16* clock cycles. 

The resistors form the weightings of an *8-tap* FIR digital filter; it just happens one of the weights is zero.   This effectively filters the harmonics off the square-wave; the FIR filter notches are at the harmonics positions.   The first non-zero harmonic is the 15th harmonic - the level is quite low.   This way you get a clean sine-wave.  The upper harmonics can be removed with an analogue filter with a much higher cut-off than trying to filter the square-wave with an analog filter.

You might find some old magazine articles under "magic sine waves" but the formulas had errors.  Other terms are digital sine waves and quasi sine waves.

---

Ah, I see your problem now.

The two *clock* inputs should be tied together and there should be a link between Q3 and Din_B  (I'm calling A Q0 to Q3 and B Q4 to Q7. Some datasheets will start at Q1 and go to Q8.)

Here's the idea, except it uses a different output filter, one with 8 non-zero taps (the 7 resistor filter works perfectly well also)
(https://www.electroschematics.com/wp-content/uploads/2009/06/digital-sinewave-oscillator-400x252.jpg)

Thanks Rob, that diagram is much more like I was expecting to see!

I understand the gist of how it's supposed to work, although I wouldn't have thought of describing it as a FIR filter (I like that way of looking at it though) but I couldn't see how the schematic previously given was going to achieve that, since the two registers seemed to be paralleled up rather than in series.

I note also your 4015 diagram shows the Resets tied low. Which is what I expect.

It makes you think did the schematic as drawn ever work.  I can't see how it can, not even in a weird way (you can never tell with these things).  Perhaps the error was from going hand drawn development notes to the computer.

Quote from: Rob Strand on June 05, 2023, 06:05:55 PM
It makes you think did the schematic as drawn ever work.  I can't see how it can, not even in a weird way (you can never tell with these things).  Perhaps the error was from going hand drawn development notes to the computer.

Yeah, my thoughts too. I alluded to this in my first post back when this came alive again:

Quote from: ElectricDruid on June 04, 2023, 06:08:08 AM
Be aware that that schematic is from a simulation. So the circuit might be tested in the sim alone, and it doesn't look entirely complete to me - or at least assumes quite a lot of stuff.

https://www.diystompboxes.com/smfforum/index.php?topic=55075.msg1268177#msg1268177 (https://www.diystompboxes.com/smfforum/index.php?topic=55075.msg1268177#msg1268177)

Quotehttps://www.diystompboxes.com/smfforum/index.php?topic=55075.msg1268177#msg1268177

Reading over the earlier posts I think the author had good intent entering the design into CAD so others can use it.  It's just that a few things went wrong.

Many thanks to both of you for helping me understand this. I'm still working on a redraft of the schematic and when it's finished I'll have to learn how to post it here. You guys are great! I really appreciate your efforts. 

Rob thanks for posting your sine wave converter. Its much better than the one on the original schematic which uses some hard to get resistor values. Can I use one section of the 4093 on your circuit? Another question. Are the 4040 divider outputs square wave? If so can I use those outputs as square wave sends? On the original schematic a switch is used to select the octave. I'm thinking about feeding the output of each octave into a channel of a virtual ground mixer to level them out.  Can that work?
What I'm trying to do is use Marcus' E&MM Harmony Generator clone as a master tone generator for a guitar synthesizer I'm designing. The tone generator would have sawtooth, sine and square wave outputs to simulate other musical instruments.     

Quote from: RGP on June 07, 2023, 11:42:15 PM
Rob thanks for posting your sine wave converter. Its much better than the one on the original schematic which uses some hard to get resistor values.

Both the 7 resistor and 8 resistor circuits work fine but there's a whole lot of finer points to selecting those resistors.  When you find these circuit you do not know what the designer has allowed for.   It's pretty normal to have funny resistors values.   Basically have to decide what how far you are going to go with the resistors:
- only E12 values (10k, 12k, 15k etc)?
- allow E24 values (10k, 11k, 12k, 13k, 15k).
- allow series or parallel resistors? (and then E12 only, or E24
- other valued resistors which are available (like 22.1k)?
- what tolerance resistors 5%, 1%, better?

There's plenty of options and you need to pin things down.

A significant factor is the CMOS 4000 gates have quite a but of output resistance and that resistance needs to allowed for when selecting the resistor, ie. subtracted off, the exact resistor values.  The gate resistance isn't 100% precise so ideally you want to dilute its effect.   If you at the original circuit If you make the resistor largish then the gate resistance has less effect on the results.   Look at the original circuit the smallest resistor is 10k and the next is 10.8k.   If the gate impedance is 0.4k  then the resistances are really 10.4k and 11.2k.  So ratio has gone from 10.8/10 = 1.080 to 11.2/10.4 = 1.077.   Whereas if you used 100k resistors the 0.4k impedance has less effect.

If you look at the original design there is a 100k and a cap following the sine-generator resistors.  You would be better off removing the 100k and increasing the sine-generator resistors and connecting the cap to that.   The effective resistance of the filter is all the sine-generator resistors in parallel, and you use that resistance to calculate the filter cut-off.

When you consider resistors tolerance that will determine how accurate you need to go in choosing the resistors.   Also, the first two smallest resistor are more important to get right than the others.

A set of accurate values which:
- allows for 0.4k of gate resistance
- emphasizes accurate of the smaller valued resistors
- raise the impedance of the resistors so the 100k filter resistor can be removed.

7 resistor: 180k, 100k, 75k, 68k+1k2 in series, 75k, 100k, 180k
8 resistor: 300k, 100k, 68k, 56k+1k6, 56k+1k6, 68k, 100k, 300k

Unfortunately I can guarantee these are *the* best resistors possible.   In the past I wrote elaborate programs which searched all the resistor combinations to find the best ones.

QuoteCan I use one section of the 4093 on your circuit?

Yes, any inverter is fine.

QuoteAnother question. Are the 4040 divider outputs square wave? If so can I use those outputs as square wave sends? On the original schematic a switch is used to select the octave. I'm thinking about feeding the output of each octave into a channel of a virtual ground mixer to level them out.  Can that work?

Yes, they are square.

Remember that the sine generator section divides by 16 so the correct tap for octaves need to be 4 Q outputs higher in number to match the sine frequency.  When you switch the sine octave-switch technically that means you should shift-up all those taps to make them keep same relative pitch to the sine.    If you want to preserve the pitch of octave relative to the sine you would be better off having a second 4040 driven by the same clock going to the sine-generator and leave the*single* switch for the clock like it is now.

Another perhaps smaller point is the sine generator requires the shift register to be initialized to all zero's or all ones.  Then once that happens the initial pattern square pattern cycles around the registers (inverted each pass).   Since you have square-wave  generators available there is another form of the digital sinewave generator where a square-wave is fed into one end of the shift register (the Din) and allowed to fall off the end.   This method is more robust since the square wave on the input is replenished all the time.  In the existing method once the shift register bits are corrupted the junk pattern keeps cycling around.  If the registers don't reset cleanly at power-up the junk (non square-wave) cycles around from the start.

You can see the subtle difference here where the shift register and divider are separate,

https://www.eeweb.com/wp-content/uploads/articles-app-notes-files-create-sinusoid-1299214389.pdf

You would need to consider how to implement the switching of the sine octave and you octave outputs with this new scheme.    I'm just mentioning it in passing.

Quote from: RGP on June 07, 2023, 11:42:15 PM
What I'm trying to do is use Marcus' E&MM Harmony Generator clone as a master tone generator for a guitar synthesizer I'm designing. The tone generator would have sawtooth, sine and square wave outputs to simulate other musical instruments.     

You should probably have a look at https://www.diystompboxes.com/smfforum/index.php?topic=122086.0 (https://www.diystompboxes.com/smfforum/index.php?topic=122086.0), then.

Rob. Thanks for all that info.  :) I'm not even sure if what I want to do is possible at this point, but I'm determined to make it work. The square to sawtooth wave converter I'm hoping to use is from R.G. Keene's harmony generator he posted here quite a while back. He used a circuit from a console organ that I'm familiar with. Early on in life I worked part time repairing console organs and came to really admire what they were capable of producing using simple passive filters to emulate instrument voices. 
   Another option for converting the square to sine is a ramp to sine converter that uses resistors and diodes, but I'm not sure what the difference is between a square and ramp wave, but I'm guessing a ramp wave has a shorter time period. I think the 4015 based converter is the better option. I'll read that app note you posted a link to carefully to gain a better understanding of it.
   Ben. Thanks for the reference you provided. as well. I looked at it once before, a year or two back, I think, but couldn't quite grasp all the techincal aspects at the time. I'll take a closer look now.             

QuoteRob. Thanks for all that info.  :) I'm not even sure if what I want to do is possible at this point, but I'm determined to make it work.

I kind of gathered that but I put it up FYI.

The 8-tap values schematic I posted earlier aren't too bad anyway.   Technically the 120k should be 130k but when you consider the gate resistance the difference is almost non existent.  The "better" resistors I posted later can shave off about 7 to 10 dB off the 3rd and 5th harmonic but they are already pretty low with the original circuit.   If you consider mixing in other "nastier" waveforms the purity isn't a big deal anyway.  (I didn't check the previous 7-tap versions although I know the values I posted won't be bad and would compete with the 8-tap.)

Quote
Early on in life I worked part time repairing console organs and came to really admire what they were capable of producing using simple passive filters to emulate instrument voices

I admire all those filters in the old organs too.  Someone did a lot of brain twisting to get those.    I always wonder what process they went through to get there.

QuoteAnother option for converting the square to sine is a ramp to sine converter that uses resistors and diodes, but I'm not sure what the difference is between a square and ramp wave, but I'm guessing a ramp wave has a shorter time period. I think the 4015 based converter is the better option. I'll read that app note you posted a link to carefully to gain a better understanding of it.

That would work.  Some of the decision points between one and the other are a fairly fine line, perhaps coming down to exactly what you want to do.   FYI with different resistor values you can actually create a nice triangle off that shift register (or other waveforms).   I'm not suggesting feeding that triangle waveform into a shaper.  I'm just saying you could create an alternate waveform off the sine-generator which you can select from a switch: then you have square, triangle, sine.

The app note is just a more robust way of doing the same thing.  You would only consider it if you have the extra divider at the input floating around for free.  (sort of FYI)

Rob. I'm sticking with the 8 tap and updating to the resistor values you posted.  ;) Thanks for letting me know the sawtooth conveter should work. Those old gents designing the console organs really had their act together in my opinion. You mention the 3rd and 5th harmonics being a bit weak. Are you refering to the 3rd & 5th intervals or the upper octaves? If its the intervals you are mentioning is there any way you can think of to boost their levesl? Oh, and that reminds me of a question I've had about the harmony generator. Can the 3rd and 5th be used together or only one at a time?
  On my harmony generator clone I'm only using the subocatve, unison,1 & 2 octaves above because few instruments are above that range. My next step is to develop circuits to emulate instruments and I'll need a trigger and gate for envelope generators. On Marcus' string pluck detector can I tap off between the diode and 555 timer trigger input for my trigger? Would I need some sort of buffer for said trigger tap? I'm thinking I'll need to add a guitar to gate circuit onto the AMS100, but not sure where to shoehorn it in. Any suggestions?
   I really appreciate how much help you've been giving to me on this project. I need to start posting schematics soon. How do I go about that?         

Quote from: RGP on June 09, 2023, 01:50:07 AM
Rob. I'm sticking with the 8 tap and updating to the resistor values you posted.  ;) Thanks for letting me know the sawtooth conveter should work. Those old gents designing the console organs really had their act together in my opinion. You mention the 3rd and 5th harmonics being a bit weak. Are you refering to the 3rd & 5th intervals or the upper octaves? If its the intervals you are mentioning is there any way you can think of to boost their levesl? Oh, and that reminds me of a question I've had about the harmony generator. Can the 3rd and 5th be used together or only one at a time?

The resistor affect the purity of the sine wave.  The 3rd and 5th harmonics I was talking about are the harmonics in the sine wave.  You can think of these as small levels of distortion of the sine-wave. 

All the divider outputs are square and of equal level.   You can do what you like with those without any special considerations.

Quote
  On my harmony generator clone I'm only using the subocatve, unison,1 & 2 octaves above because few instruments are above that range. My next step is to develop circuits to emulate instruments and I'll need a trigger and gate for envelope generators. On Marcus' string pluck detector can I tap off between the diode and 555 timer trigger input for my trigger? Would I need some sort of buffer for said trigger tap? I'm thinking I'll need to add a guitar to gate circuit onto the AMS100, but not sure where to shoehorn it in. Any suggestions?

Yes, that point is a good place to tap off.   To get a clean trigger you can connect an inverter (or two inverters in cascade) to that point.   The trigger at the NE555 input is a negative going pulse, approx.  So one inverter will convert that to a positive pulse.   With the 10k and 10n parts the trigger pulse width is about 40us (0.4*R*C) as seen by the NE555.  A CMOS gate will see a little longer than that, say 70us (0.7*RC).

You might even be able to take the outputs off the existing 4093's.   Points along that part of the circuit have different timing but it might still be usable provided you get the pulse polarity correct.

Quote
   I really appreciate how much help you've been giving to me on this project. I need to start posting schematics soon. How do I go about that?       

No problem.   To be honest there's a lot of other people here that play with these types of pedals more than me.

Below the text box when you post is an "Add image to post" button.  You can use that to attach images - it actually automatically links to the real image and puts a thumbnail in the post.  The service has been a little unreliable lately if it doesn't work try again later.   (It might be wise to copy and paste your post text into a document and save it on your computer, just in case something goes bung.)

Rob. You may have seen Ray Marston's ramp to sine wave converter circuit that I mentioned before. This circuit seems too simplistic to actually converet to a sine wave, but if Ray based the circuit on old console organ circuits it may indeed work. What I'm wondering is can it convert more than one frequnecy being fed into it. I think I may have posted this, but I'm not sure if it worked.   
(https://i.postimg.cc/S2gnCt1z/marston-ramp-to-sine-wave-converter.png) (https://postimg.cc/S2gnCt1z)

Ah, good the image is there. Maybe this simple approach may workm but I have my doubts that it will work for multiple frequencies. What about dupicating it for each octae I'm using?   

There's no frequency-dependent parts in that (except the opamp will struggle above the audio band). In concept it works "to DC". (And is much older than Marston: I've seen it with vacuum diodes, maybe in Korn&Korn (https://historictech.com/product/electronic-analogue-computers-book-korn-korn1st-edition/)?)

The drawback is that the waveform has glitches, high harmonics, which are easy to hear. The classic analog synths used BJT diff-amps which are smooth to a point, but that point never goes away. The diode-thing is often used in function generators to make a perfectly good sine-like wave.

Ahhh.... and these techniques start from a TRIANGLE wave, not a RAMP.

PAIA-guy John had a 50 cent trick for making a ramp (the easiest wave) into a triangle, but it was fussy and glitchy. Basically a BJT "cathodyne" overdriven so half of the wave was inverted and the other half just bled-through raw.

Hello PRR. I've read many of your posts and really learned a lot from them. You have a talent for explaining things that makes it easy to understand. Thanks much for that. The Marston circuit is calling for a ramp wave at it's input. Could a square wave work for his circuit?

Quote from: RGP on June 10, 2023, 02:09:45 AM
The Marston circuit is calling for a ramp wave at it's input. Could a square wave work for his circuit?

No! Absolutely not!

It works by taking a linear shape and applying carefully calibrated distortion to it (what we'd call soft clipping) to turn the waveform into a sine. I *know* it say "Ramp" by the input, but I agree with PRR that what it *actually* wants is a triangle wave.
If you put a square wave into it, you'll get something very much like a squarewave out as well.
The circuit will convert any *frequency* you feed it, since it's just applying distortion, so it's not fussy about that. What it *is* fussy about is *amplitude*. If the input signal isn't exactly the right level, you'll either get not enough distortion or too much, and the output won't be a sine wave. That means that these kind of circuits are pretty good added to a basic VCO core to provide more waveforms where you have a fixed output level, but they don't work in all situations.

Turning a ramp into a triangle wave isn't hard - you just feed it through a full wave rectifier.

I've got some examples of this sort of stuff in a synth context over on my site, if you're interested:

https://electricdruid.net/a-study-of-sub-oscillators/

Thanks Tom. Your article has me understanding the wave forms better. I'm new to this, but not to electronics.   

Where is the full Marston article?

Quote from: PRR on June 10, 2023, 01:59:32 AMPAIA-guy John had a 50 cent trick for making a ramp (the easiest wave) into a triangle.....

https://modularsynthesis.com/paia/4700/4720A/4720a.htm
https://modularsynthesis.com/paia/4700/4720A/PAiA%204720A%20Voltage%20Controlled%20Oscillator%20Manual.pdf
https://modularsynthesis.com/paia/4700/4720A/PAiA%204720A%20Voltage%20Controlled%20Oscillator%20Schematic.pdf

The manual explains the ramp-to-triangle gizmo on printed page 19.
The manual has a cut-off schematic.
I remember staring at a 'scope, AND listening, for an hour, and realizing it doesn't get better than this. (Actually it did, on our ARP, but that was >1000X more expensive. And IIRC, it made BOTH the ramp and the tri in two separate integrators with a trick /2 current source.)

Note the diode shaper to make sine from triangle. The 2-transistor scheme would have been knowable by John but bulk-buying diodes (and outsourcing labor to kit builder), the diode plan may have been cheaper.

Hey Paul,

You mentioned the tri-to-sine waveshaping that never removes the "point". This is often known as the "cusp" or "pip", and there's been a lot of effort spent trying to get rid of it. Tim Stinchcombe has a lot of good references and a nice overview:

https://www.timstinchcombe.co.uk/index.php?pge=trisin

The OpenMusicLabs paper about it isn't bad either:

http://www.openmusiclabs.com/files/otasine.pdf

Hope you find something interesting amongst it,
Tom

QuoteAh, good the image is there. Maybe this simple approach may workm but I have my doubts that it will work for multiple frequencies. What about dupicating it for each octae I'm using?

The main advantage of sine-shapers is there is no filtering and they operate over an enormous range of frequencies.   Function generators with square-tri-sine outputs quite often use sine-shapers.   The purity of the sinewave isn't that great, say 1% THD.   For applications with limited range or fixed frequencies you can improve the purity with a fixed filter.  For your application I doubt anyone would know if you filtered off everything above 2kHz.

QuoteYou mentioned the tri-to-sine waveshaping that never removes the "point". This is often known as the "cusp" or "pip", and there's been a lot of effort spent trying to get rid of it. Tim Stinchcombe has a lot of good references and a nice overview:

Yes, the sine shapers which use the "differential pair" (like the example on your site) and the diode type used by Marston have pointy tips.   To some degree this is a sign the high order harmonics don't roll-off very quickly.

The types of shapers that use a diode limiters produce aesthetically pleasing sine-waves without tips.  They also roll-off the high spectra better.   The 80's era function generators often used this type.   The THD is still about 0.5% to 1% and with some effort you can sometimes null one of the lower order harmonics.    These sine-shapers look more complicated because each break-point requires four diodes instead of two (like the Marston version).

Sine-shaper is top left section:
(https://i.postimg.cc/zbVH8mFG/GFG8016-D-2-sine-square.gif) (https://postimg.cc/zbVH8mFG)

At some point I found this one which is a little more economical.  Quite good output.  Input needs to be 4.0V to 4.1V peak but you could tweak the input divider for different input levels.

(https://i.postimg.cc/c6cG0K66/triangle-sine-shaper-1132353-maybe-ED-mag.gif) (https://postimg.cc/c6cG0K66)

Fairly impressive performance, especially for a simple circuit:

(https://i.postimg.cc/7fdkQFfN/sine-shaper-ED-waveform.png) (https://postimg.cc/7fdkQFfN)

(https://i.postimg.cc/GB11bvP8/sine-shaper-ED-spectra.png) (https://postimg.cc/GB11bvP8)

---

I'm still trying to find the origin of this circuit.  I could not find it previously.
One reference is from Wavetek's 1983 patent.  Except here it is used in reverse.
(Wavetek are famous for there function generators.)

https://patentimages.storage.googleapis.com/f5/8f/6a/adbe36c8d96aef/US4415860.pdf

EDIT:
- I have confirmed this form of the circuit was used on Wavetek Function generators after the mid 70's.
- The example circuit above connects the second set of diodes to ground whereas the Wavetek
   circuits connect this point to the input terminal of the first pair of diodes.

---

I've some stashes of info on sine-shapers. Here's another one which came to mind,
https://till.com/articles/sineshaper/

PRR. It's in Marston's book. Here it is.
(https://i.postimg.cc/3dsTxxdz/Marston-s-110-Op-Amp-Projects.png) (https://postimg.cc/3dsTxxdz)
Rob & Tom Wow! What a wealth of info. Thanks for shairng it. 

PRR. If you don't have a digital copy of the Marston book I'd be more than happy to e-mail it to you. Since the circuit needs a triangle wave input that makes it easiser for me to choose a square to triangle wave converter and gives me a 4th waveform to boot. I found this one earlier today.   
(https://i.postimg.cc/YLNs2MZB/Screenshot-2020-07-24-Elektor-301-Circuits-79-179-pdf.png) (https://postimg.cc/YLNs2MZB) The others than Rob & Tom posted arer also great choices too.

I've redrafted Marcus' schematics and added notes about setting the trim pots. I drated it all onto a single page, but this is likely too large to post. I'll break it down into sections & post them soon. The big question that still remains is where does the input signal come from to feed the EMM harmony generator clone? I suspect it's from the AMS100's compressor output but can't confirm that as a fact. Marcus' schematic also states some sort of filtering was used, but only he knows what it was.     

Quote from: RGP on June 10, 2023, 10:55:26 PM
PRR. It's in Marston's book. Here it is. .... a digital copy

https://archive.org/details/110OpAmpProjects/page/n81/mode/2up?view=theater
like print page 77

The ARP 2500 had over-engineered VCOs and sine-filters; Pearlman was a heavy geek.
http://www.guitarfool.com/ARP2500/1004.html

Quote from: Rob Strand on June 10, 2023, 07:19:50 PM....diode limiters produce aesthetically pleasing sine-waves without tips....

Typically by over-driving them, widening the wave, raising the 3rd and 5th. Nice phatt sound, but something I was doing didn't want that. Forget what it was.

Altho 3rd -90dB 5th -50dB aint bad.

And yes, there may be much else from the last 40 years I have missed.
__________

> the origin of this circuit.  I could not find it previously.  One reference is from Wavetek's 1983 patent

As said, the technique goes way back to hollow-state days.

Electronic Analog Computers, 1956,  by Granino Korn and Theresa Korn
https://archive.org/details/in.ernet.dli.2015.84794/page/n309/mode/2up?view=theater
Print pages 290-299 show vacuum diode wave-benders of every sort except sine, "many combinations are possible".

Maybe Theresa never published a good derivation of a sine shaper? (Maybe she held it as a commercial secret?)

PRR. I think you'd need a pilot's license for the ARP :)

A copy of Operational Amplifiers: Design and Applications by Jerald G. Graeme at the bedside is mandatory:
http://www.introni.it/pdf/Graeme%20-%20Applications%20of%20Operational%20Amplifiers%203rd%20generation%20techniques%201973.pdf

I see there is a later "3rd generation" edition.

QuoteThe ARP 2500 had over-engineered VCOs and sine-filters; Pearlman was a heavy geek.

It's not over engineered. It's got deliberate motives.  The first two diodes are the bare minimum "diode clipper" triangle to sine, which bends lower voltages.  Then the 4xdiode rounds the peaks to remove the pointy tip issue.  For the diodes it uses transistors for better matching so the peaks are accurately shaped.  I kind of like it.  I got complex shaper circuits which are very ineffective and probably not much better!

Quote> the origin of this circuit.  I could not find it previously.  One reference is from Wavetek's 1983 patent

As said, the technique goes way back to hollow-state days.

The Wavetek one is a unique connection which seems to work well.  Tricky stuff like this is where people like Wavetek and HP (Agilent etc.) stayed ahead of the competition.   (If you plot the derivative of the output it is very sine like. where on some poorer ones the derivative looks like a square wave.)

The classic connection is the Marston type where the diodes tap off a divider.   That's the form most books.
IIRC the shapers in the ICL8038 chip sort of follow the classic structure and replace diode with transistors; XR2206 apparently uses a diff amp.

Then there's circuits which only use the 4xdiode structures like the Goodwill function generator I posted earlier.
(There's some twists on this one.)

There's a whole heap of patents with differential amp based circuits.   HP have a patent on at least one.  I don't know what product it is used in.

QuoteAnd yes, there may be much else from the last 40 years I have missed.

It's impossible to keep up with this stuff.  There must be hundreds of function generators and synths out there with shapers in them.   Today it's almost obsolete technology.

I finally have the schematic redrafted. It is in 3 parts. The 1st is the modified AMS100 input processor. The 2nd is Marcus' pluck detector. The last one is The E&MM Harmony Generator with Marcus' mods and mine, so far. 

HMM, somtthing went wrong with posting the images. I'll try again. I'm gertting a message no images can be uploaded. Is this a common problem?

Trying again to upload. Sorry, still not working.

Quote from: RGP on June 11, 2023, 11:05:46 PM
Trying again to upload. Sorry, still not working.

(https://i.postimg.cc/XGSw3khy/100-Hzmoderatehum.gif) (https://postimg.cc/XGSw3khy)
Not relevant, but posting. How are you doing it?

PRR. I'm trying to add the image to the post. It didn't give me any problems before. Is there a better way?

Trying again. Could it be the file extension or size causing the problem?

You might like this one. 
(https://i.postimg.cc/BX2KW0sC/organ-voices.jpg) (https://postimg.cc/BX2KW0sC)

trying again. No luck

Trying with a different file extension
(https://i.postimg.cc/7JCGB5nx/AMS100-Input-Processor-with-Marcus-W-Mods-01.jpg) (https://postimg.cc/7JCGB5nx)

My file size was too large. JPG works. Here's the E&MM Harmony Gen. Clone
(https://i.postimg.cc/w1Fqm6DF/E-MM-Harm-Gen-with-Marcus-W-RGP-Mods.jpg) (https://postimg.cc/w1Fqm6DF)

Hetrer's the pluck detector
(https://i.postimg.cc/yW97ThFD/Marcus-W-String-Pluck-Detector-01.jpg) (https://postimg.cc/yW97ThFD)

Ops, more bad typing. Sorry. The images seem a bit blurry but legible. If there are any issues seen in these let me know so I can correct them

Now another question. On the Harmony Generator. Do I need to add some sort of buffer on the square wave output mixer to split it into two dedicated outputs? One for a square wave send and the second to a square to triangle converter.

Quote from: RGP on June 12, 2023, 01:01:24 AM
Do I need to add some sort of buffer on the square wave output mixer to split it into two dedicated outputs?

If you mean the output marked "Square Send" then the answer's "No" because it's already coming directly out of an op-amp, and that op-amp can easily drive two inputs.

Thanks for clearing that up for me Tom. Did you see any errors with the schematics? 

Quote from: RGP on June 12, 2023, 12:53:05 AM
Hetrer's the pluck detector
(https://i.postimg.cc/yW97ThFD/Marcus-W-String-Pluck-Detector-01.jpg) (https://postimg.cc/yW97ThFD)

What? The actual pluck?



I'll show myself out.

I found this (again), on an old and slow Chromebook, not sure it goes here, but someone will recall where we were talking digital counters to sine-step waves.
Google  digital_sinewaves_11_76
https://www.google.com/search?q=digital_sinewaves_11_76&rlz=1CAHKDC_enUS965&oq=digital_sinewaves_11_76
Find:
https://www.tinaja.com/glib/rad_elec/digital_sinewaves_11_76.pdf
Don Lancaster, {EDIT} so it is to the point and correct.

Early patent for the conventional diode tapped sine shaper (1951/1956),
https://patents.google.com/patent/US2748278A/en?oq=+us2748278

QuoteDon Lancaster, so it is to the point and correct.

I like Don Lancaster's stuff but that article is full of errors.

In your second link check out the table and the 16x clock values.  He is using the 7-tap (vs 8-tap) version (in fact he uses the type which always has an unconnected tap; ie. zero filter weight).

Tell-tale signs something is wrong:
- The single value resistor tap is the high-valued resistor where it should be the lowest valued resistor.
- The resistors that are close in value are the high valued resistors instead of the low valued resistor.

The root cause is the ratios in parenthesis (and the formula) are for *conductances* not resistances.
16x clock, 7-tap: 
incorrect resistances: 22.1K; 41.2K; 53.6K; 57.6K; 53.6K; 41.2K; 22.1 K
conductance ratios: (1.000)(1.849)(2.412)(2.613)(2.413)(1 849)(1.000)
invert conductances to get resistance ratios
  (1.000)(0.5408)(0.4146)(0.3827)(0.4146)(0.5408)(1.000)

As a crossreference, the book Art of Electronics by Horowitz and Hill  gives a 16x clock example with 7 resistor taps  using a CMOS 4015 device.
The resistor values are:  57.6k, 30.9k, 23.7k, 22.1k, 23.7k, 30.9k, 57.6k

The resistance scale factor 1 corresponds to resistor value 57.6k,
corrected resistance scale factors: (1.000)(0.5408)(0.4146)(0.3827)
resistor values approx 57.6k time these: 31.2k, 23.9k, 22.0k

We can see the corrected calculations, based on conductance ratios, agrees with the Horowitz and Hill resistors,
within rounding to the nearest (funky) resistor values.

Exact values for 22.1k: 57.75k, 31.25k, 23.92k, 22.10k, 23.92k, 31.25k, 57.75k

There were other errors in the article.  I'm not even sure the formula produces the correct ratios, or that the example circuits have the correct resistor values.  I have gone over this subject at some point.

Thanks for the info. I'll need to study it. Do either of you see any errors or problems with the 3 schematics?  I'm still looking at the Elektor square to triangle wave converter. Unfortunately I can't find the text for that schematic.   

The point labeled as U2. What is it? Every PDF I found for 301 circuits doesn't include that circuit or it's related text.   

QuoteThanks for the info. I'll need to study it. Do either of you see any errors or problems with the 3 schematics? 

I got a security issue trying to look at the original PDF to compare it with.   Haven't looked at the cause. yet.

QuoteI'm still looking at the Elektor square to triangle wave converter. Unfortunately I can't find the text for that schematic.   

Elektor?  which one?   You mean the the circuit I posted here?

https://www.diystompboxes.com/smfforum/index.php?topic=55075.msg1268263#msg1268263

QuoteThe point labeled as U2. What is it? Every PDF I found for 301 circuits doesn't include that circuit or it's related text.   

Still not sure what circuit you are referring to.  Which post did it appear?

Rob. It's posted here somewhere, but I'm attaching it anyway.
(https://i.postimg.cc/TytXbNLP/Screenshot-2020-07-24-Elektor-301-Circuits-79-179-pdf.png) (https://postimg.cc/TytXbNLP)

Rob. I have the pdf onmy thumb drive. Should I attach it here?   

Quote from: RGP on June 16, 2023, 10:33:07 PM
Rob. It's posted here somewhere, but I'm attaching it anyway.
(https://i.postimg.cc/TytXbNLP/Screenshot-2020-07-24-Elektor-301-Circuits-79-179-pdf.png) (https://postimg.cc/TytXbNLP)

Circuit from:  Elektor May 1978, p5-46 to p5-47

(Text translated from German Issue)
"Figure 3 shows a circuit that compensates for the decrease in amplitude with increasing frequency by increasing the charging voltage U2. The function of the short-circuit switch in FIG. 2 is performed here by transistor TI. Its collector-emitter path becomes conductive at the time of the positive edge of the square-wave input signal, so that capacitor C3 is bridged for a short time. On the falling edge of the input signal, T2 conducts; at this point, C4 is charging. The mean value of the current is proportional to the frequency of the input signal within a specific frequency range. So if the input frequency increases, the voltage at C4 also increases practically linearly. The result is an exponential sawtooth voltage with an amplitude in the range of 60 Hz. . . 10 kHz remains constant; however, the shape of the sawtooth still depends more or less on the frequency. This blemish (by the way, a linear sawtooth has less "musical content" than a sawtooth with an exponential rising edge) can be eliminated by replacing resistor R4 with a current mirror (T3/T4 in Figure 4). An output buffer (T5) completes the square-wave sawtooth converter ..."

I believe the idea is that as the frequency increases the DC level at U2 increases to compensate for the drop in level at the main output (the exponential sawtooth).   I have not checked how well this works.   That idea applies to both Fig 3 and Fig 4.

QuoteRob. I have the pdf onmy thumb drive. Should I attach it here?

AFIK, can't attach PDF's or can only attach small ones.

Rob. Thanks for explaining the circuit to me. I'm leaning toward using it. Hope you can figure out why the pdf doesn't show up. I read Marcus' posts over many times to get the schematics right, or at least hope they are. Still wondering where the input signal is coming from and the filter he used. I'm thinking the AMS100's compressed output. Does that seem right to you?     

Quote from: RGP on June 17, 2023, 01:54:31 AM
Rob. Thanks for explaining the circuit to me. I'm leaning toward using it. Hope you can figure out why the pdf doesn't show up. I read Marcus' posts over many times to get the schematics right, or at least hope they are. Still wondering where the input signal is coming from and the filter he used. I'm thinking the AMS100's compressed output. Does that seem right to you?   

The PDF issue is just a limitation of the forum.  It stops people uploading crazy large files - which would probably clog-up the database.

Yes, the input comes from the AMS100.   I had to decipher what was going on there early on in the thread.

I made these short notes at the time.

---

"This project uses part of the AMS100 circuit from DEVICE magazine.
In particular the envelope extractor from part 1.

The schematic starts at the peak detector output. The buffer on the
schematic is a replica of the buffer on the AMS100.1 schematic.
The one after the peak detector.

On page 4 of the thread some mods are given."
[the thread is this thread.]

---

AMS-100 guitar processor
Appeared in DEVICE magazine

https://hammer.ampage.org/?sort=new&page=11
https://hammer.ampage.org/?sort=new&page=12

Issue 1 - part 1
https://hammer.ampage.org/files/Device1-1.PDF

---

The other parts of the AMS100 can be downloaded from there also - compliments of Mark Hammer.

Rob. I based all the schematics on that same criteria you pointed out. I have all the device issues & the E&MM Harmony Generator thanks to Mark Hammer having them on his site. I'm wondering if it's possible to use the 3rd & 5th intervals at the same time? The note hold feature Marcus came up with  is very cool, too. The sound clips he posted seems to prove the circuit works well. This, at least in my opinion, is a great start for building a monophonic guitar synthesizer. Oh, by the way, what do you think of that organ voice schematic?