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Donovan

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Everything posted by Donovan

  1. Col- What this thread DOES need is organization. Why not have one post that you repeatedly edit with the most up to date, clostest thing to success that each of you have. When you make progress, simply come back and edit it. Make it all inclusive, so that all you have to do is point us less technically adept folks to it and let us sink or swim. IMO, PSW should have done that at post #1, but why not start now and help cut down on the chapters of needless reading you mentioned? If it's well-written and coherent and speaks to the intended audience (those of us without a PHD in quantum mechanics), it would be very well received. Write it and they will come.
  2. Actually, can't count me in that list... I consider my project a failure on hold at the moment. I burned myself out from messing with that circuit and probably won't try another sustainer circuit until I learn A LOT more than I know now. I've been reading this thread on a regular basis, hoping to get up to speed, but Col and Hank are getting into some unchartered waters. With any luck, they'll have it sorted before I break down and buy a Fernandes or Sustainiac system.
  3. I've been in electronics manufacture for the last 10+ years and currently function as electornic assembly technical expert and trainer. Flooding and wicking might get the job done, but I would advise against it and use the paste and hot air technique or look into getting a tip designed for drag soldering (it holds liquid solder in a reservoir that dispenses a small amount of solder upon contact to each pin) as every time you heat the component and the solder joint, reliability of both component and PCB are reduced. One, because you increase the thickness of the intermatallic layer (junction of the solder joint and the PCB land) which means embrittlement that leads to fracturing and two because you're applying another round of extreme heat. Once you start using some of these more elaborate devices, you guys should also start considering an ESD-safe workstation to minimize the chance of rendering your devices nonfunctional due to electrostatic discharge. The basic setup would consist of a dissipative mat with a path to ground which is also common to any tools being used and the individual via a bracelt and ground strap. In general, the device datasheets should indicate an ESD sensitivity level. If it references ANY level, you should take heed in order to not waste your devices. The industry spends a lot of money on protection from electrostatic discharge prevention and this increases avery year as devices get smaller and more complex (weaker with regard to static shock). Nothing worse than trying to troubleshoot something that comes down to internal device failure due to an ESD. Checkout www.esda.org and give S20.20 a read if interested. I can also recommend very cheap ways of setting up the workstation for this.
  4. Hello gentlemen. I've recently been doing some reading about pulse width modulation using 555 timers as a means of more efficiently controlling electrical motors. The duty cycle is modulated, rather than pushing 100% of the time, it pulses power in the form of a square wave at a (for the most part) undetectable frequency. Just a shot in the dark, but could the analogy of the cruise control be realized as an AGC through the application of PWM to the output stage's power supply (or somewhere else in the signal chain) through a low frequency, say under 20Hz, with the signal amplitude modulating the duty cycle? That's pretty much the path I'm taking by using a PIC in my sustainer circuit. The PIC 'monitors' the sustainer's preamp output level & ultimately adjusts the duty cycle of its own PWM output stream to suit - if you feed this PWM stream into a low pass filter, you end up with a DC level - this DC level is applied to a 'gain control' JFET in the preamp (which adjusts the gain applied to the incoming signal to ensure a constant predefined output into the power amp) Thank you for that excellent explanation. I feel like I just learned something important and I'm glad that what I suggested had some merit, even if it's not an original idea! That is brilliant, the running it through a low pass filter portion. Am I correct in assuing this gives a nice smoothing effect, so the AGC doesn't act as choppy? Is this post-filtering a common technique and if so, what is it called or what other applications might I find it mentioned?
  5. Hello gentlemen. I've recently been doing some reading about pulse width modulation using 555 timers as a means of more efficiently controlling electrical motors. The duty cycle is modulated, rather than pushing 100% of the time, it pulses power in the form of a square wave at a (for the most part) undetectable frequency. Just a shot in the dark, but could the analogy of the cruise control be realized as an AGC through the application of PWM to the output stage's power supply (or somewhere else in the signal chain) through a low frequency, say under 20Hz, with the signal amplitude modulating the duty cycle? Forgive me if this is a daft idea for reasons beyond my knowledge level... just came to me when I read that quote. If the power amp stage would be too difficult, the signal could be sent to ground using PWM, but again, modulated, not a 100% shot at some threshold as has been recently discussed and tossed out. I do realize this is something like the "shunting to ground" that has been talked about already on this page, but was just thinking this would be a more translucent effect, in theory, or at least intuitively within the confines of my flawed psychi.
  6. Please don't be discouraged. I appreciate what you're doing. Though it's slightly over my head I read regularly and enthusiastically. I look forward to reading more of your results with this new circuit and seeing a preliminary dwg when you're ready. I've been keeping quiet here (as I am sure many others are as well) as I "learn" some more basics and tinker.
  7. Nice job! The front is very cool.
  8. Does alnico have a similar appearance/feel as ceramic magnets or is it noticeably more metal-like? Perhaps. If only we could see this stuff! Not sure I understand what you mean. I don't see the fizz as at all related to the signal being used to drive the coil. Should I? I have been able to drive the thicker strings quite hard, but the B and E strings are not effective above about the 14th/15th fret. Also, I have never gotten a consistent fundamental mode or harmonic mode. I have been able to use 2nd order all-pass filters to shift the regions and harmonic around, but there is always a mix in some places, harmonics in some places, and fundamental in others, never just one solid mode everywhere. This is 100% consistent across multiple drivers, multiple circuits with different preamp and amp configurations and different host guitars. Always, I am accompanied by a good deal of fizz. When I dial the amp circuit back to where fizz is gone, sustain loss abruptly follows. The problem has been consistency and strength on the B and high E strings above the 14th/15th fret. I have completely removed the middle pickups with no change in results vs. leaving the pickup installed and connected. Quite the opposite actually, I was able to get the same results using the middle pickup for the signal as the bridge pickup. I was originally using a test box with about a 2-3 foot in/out cable. All of my electronics were contained within a black box and I was building modular preamps, filters, and amp circuits as to easily connect/disconnect for testing. The results with this box were the same and when I brought my less than ideal results here, it was suggested then to ditch the box and do a direct install, biting the proverbial bullet. Again, the results are the same. I have fiddles with preamp gain, no preamp gain, 20X 386 gain to 200X 386 gain, driver distance to strings, signal pickup pole distance to strings, low pass filtering anything above 2kHz, all pass filters, half wave rectification, injecting all sorts of digital effects into the signal chain, all to no avail. I am at a standstill at the moment and have even considered biting the bullet and buying a sustainiac or fernandes system, if for nothing else to dissect the thing and make it all public (at least this should be simple enough for the driver portion) once and for all. You have consistently pointed to my driver being the issue over and over and I am convinced it is the biggest part of my problem. Hopefully, it'll get sorted. So, until I can build another driver... You guys are brilliant. Keep it up. I am still hopeful.
  9. Here I go again being intuitive. Let's call this extra length of magnet that hangs down "extra" for the moment. This extra length, is it being used? Is it not sending more magnetic flux laterally outward than shorter magnets of equal strength? You have gotten handy with some software that models fields, correct? Is this something you could model easily enough? Regardless, it sounds as though I am completely off base, but do you see where I am coming from at least? I mis-worded that. I want to not have the field so lopsided as it seems it would be in the configuration I have. Instead of equal magnet, I should have said equal field, as I would not expect that having much if any actual magnetic material above the coil is of use as it will put too much constant pul on the strings and at some point hinder any vibration. Isn't the entire field a sum of two components, a permanent field plus a temporary field's shift in one direction or another? So, whereas the overall summed effect is that the permanent field is weakened or made stronger, isn't this due to the continual destruction/reconstruction of the temporary portion in opposite polarities? Are you certain that fizz is due to the the strings acting as flux conduit? Brings me back to my original question about the extra magnetic material... I guess I feel fizz could be coming from the permanent magnets hanging down and emanating flux laterally. I never got fizz in the tests that I did holding the driver above the strings, upside-down. It only began occuring once the driver was actually installed to the pickguard. I then thought that some of it was due to shielding on the pickguard itself as Sustainiac removes the pickguard shielding, but then I installed the driver on my guinea pig strat, which is not shielded except for a tiny section, and the fizz is still as evident. More of the intuitive. I have tried only pulling, figuring the temporary portion of the field would react quicker and it would be a power savings as well. However, testing I did was completely unsuccessful. If I had to guess why, I'd say the coil could be likened to a speaker, in that it can not reproduce detail perfectly and so rounds any sharp changes, much like a speaker rounds the corners of a perfect square wave. I have no reason other than I guess I've grown discouraged and have been waiting for some materials to try something drastically different, but now that you've suggested it, I may. Thanks for the discussion.
  10. I didn't understand that, can you try and explain again ? I've been using SC strat bobbins. Not the cheapo type with metal poles and a bar magnet (though I have a few of those as well), but rather 100% magnetic pole pieces. My construction has been: unwind stock PU, bock up all but upper 3mm, wind coil in upper 3mm. This leaves the lower remainder of bobbin empty except for the lower portions of the pole pieces (total height minus the 3mm coil). I've not measured it, but I'd guess it's around 7mm to 10mm. What is the effect of having these poles left in their entirety as opposed to having polepieces only within the 3mm bounds of the coil? I know this is a theoretical best guess type of question at this point, but intuitively, I would think there is some waste with the extra length or an increase in fizz or some other EMI-related philoso-babble. I add this as question number 379 under "sustainer questions that shall never be (other than empirically) answered". Intuitively, I would think the system would like it better to have an equally-sized permanant magnetic field both aboe and below the coil. I would also think that as is, there would be more phase lag or shift on one half of the AC wave form than the other as there must be some sort of effect on destroying and rebuilding the field in the opposite polarity.
  11. Just checking in. I've stopped work on this project as I need a GOOD driver and I don't think my attempts have been there as of yet. I've used up my stock of spare pickups. No data, but something tells me that the extra pole magnet that sticks out from the used up 3mmm of the bobbing adversely effects the sustainer... any ideas? I have read here numerous times about not attempting to cut down neodymium mags, but what about ceramics? Can one safely file thsese down? I feel I need to make a bobbin from scratch, which IMO is a larger undertaking. I want to try and take advantage of multiple designs by making a dual coil thin design. I have my pals at work keeping their eyes out for scrap material for me to play with and until then, it's playing with circuits for now. Recently built and installed a 9v LED clipper fuzz on my guinea pig strat... not a permanent mod, more of an exercise in learning how to work with op amps more efficiently. Anyway, some cool posts as of late. Can't wait to see what comes of the things being discussed recently... Col's current mode 386 and hank's hex work. Keep it up gents... and pete, should you need an extra few bucks, would love to purchase a coil from you.
  12. I've considered them, but I want to use a battery for this project. The extra op-amp and two resistors that are needed to setup a virtual ground, and the very few extra components elsewhere are a bargain to pay for being able to use a single 9v battery IMO. The LM324 in the upper left of your schematic (this one) is the virtual ground, correct? It looks like you referenced the input signal to this after the 220nF cap, right? It is OK to reference as many other op amp inputs as you want to this as well? I thought they would interfere with one another somehow and so what I've been doing is referencing each op amp to half supply using the two resistors and two caps where you've used the one to send transients to ground. The results are LOTS of components, which is why I inquired about using two batteries. Space issues aside, if you were to use two 9V batteries instead of the op amp virtual ground, then theoretically, doesn't it become possible to remove not only the signal biasing, but the input and output capacitors as well, alleviating most of the circuit-induced phase issues? I'm a little hung up about understanding the difference between a virtual ground setup at 4.5V vs what I've read can be called a "true" AC when you tap the center of two seris-connect 4.5V supplies, whether they be batteries or whatever. I have trouble seeing the midpoint of two series-connected 9V batteries as being somehow different than tapping one 9V battery using a resistor network. The sources I believe are trying to say that the center point of 2 batteries is actually 0V, where intuitively I would see it as 9V. Can you comment/enlighten on any of this?
  13. Yes, I do like the versatility that uber-switching makes possible. I would like to learn more about them, any online site suggestions? I would like to read a simple site that goes into basic guitar wiring, especially switching and tone/volume controls. Uou seem to be the switch-king around here. The S-1 switch on my strat is intimidating to say the least and I would like to rewire everything neater than stock, but am at a complete loss... and yes, I've looked at the drawings on the fender.com site, but they are mfg drawings, not really a good schematic/dwg mix... sorry to get OT. Col- Have you played with dual power supplies at all to get rid of all the half power supply biasing and most of the capacitors that a single supply makes necessary?
  14. I'm still checking in regularly. I have concluded I need a better driver and so I picked up a bunch of tiny neodymium and ceramic magnets the other night to experient with. I have some steel flatstock to use for a blade and I am in-process with making a nice bobbin, but I am ditching the whole "use a pickup" idea for good. I did do some testing last week injecting lots of different effects, one by one, into the signal chain to feed the LM386 circuit and see if anything positive would happen. I fed reverb, delay, compression, chorus, phaser, flanger, tremolo, rotary, wah, auto-ya, and envelope filter in, but nothing at all got positive results. Chorus did give a pulsing effect to the strength of the sustain, but nothing useful really.
  15. Col- What do the little red X's mean on your schematic with regard to the power input/outputs on the opamps? I assume you are connecting power to them in those locations, or do you not have to?
  16. Hello all- The lead in those clips shows very nice sustain, Pete. Nice work. I think I understand what Col was getting at... just that your results may not be indicative of what us noobs can expect to get if you are not using the "standard circuit" shown here, even if you do not endorse said circuit. We need SOME circuit to use as a reference for expected results. Regardless, the past few pages of posts have prompted me to continue sitting, waiting, wishing... so something came to me, a test to try. Once it occured to me to try it, I felt like a dumba$$ for not having tried it before... testing the sustainer circuit sans the preamp and listen strictly through the wood. Is my thinking right that I should not care about loading the pickups if I am not connecting the guitar output to anything? Or would the loading still affect the loop in a negative manner? If not, then some testing last night convinced me that my driver needs work as the phase issues are still present, even sans pre-amp. Amp circuit is standard LM386 (but -N4) with only some extra power filtering in the form of bypass caps, which to my knowledge should not be the cause of any phase issues as it is not part of the signal chain. This driver is a standard strat neck pickup, originally a Fender Tex-Mex custom staggered. The bobbin was blocked off with fiberglass strips at the top, leaving a 3 mm gap and wound with 0.2 mm polysol wire to 7.9 Ohms, potted with urethane and vacuum degassed. I was VERY careful in its construction. The pole magnests have been lowered flush with the top of the bobbin. The problem I believe at this point is relative to 2 items: 1) The stock pole magnets for 2 reasons... A) although I have no evidence, I think I may need to remove these in favor of smaller, disc-shaped ones, such as the craft magnests I have heard mentioned here previously. I am thinking the extra length sticking out of the bottom of the driver is at least a potential issue for excess EMI and/or phase issues. I also am not entirely clear on optimizing magnet position in the Z axis. Should the magnets sit entirely below the coil, partly below, or span the entire depth? I am not sure if these are ceramic or AlNiCo magnets. Being as I do not care about ruining them, should I try cutting or filing/shaping them? This pickups polepieces do not align directly beneath the strings, as if the pickup were designed for the bridge position or another guitar altogether. However, I know this did originate as a neck pickup as the stock bridge was a full-size HB. Also, the original middle pickup seems to be the same geometry. Is this common or did I get the wrong pickups stock?!?!?! I have a bunch of scrap ferrite strips used in transformer lamination. The pieces are small enough that I couls layer them to create a blade or fill in the holes between the pole pieces on the bobbin. Does anyone think I should attempt one or the other? That was supposed to be a "B", not a stupis smily sunglasses guy... doh.
  17. OK, some I'm understanding the whole kink thing. I need to process it for a bit. I am ordering a cheapo Squier strat as a guinea pig guitar tomorrow as well. I have been wondering if my full size JB might be a negative in any way. Might there be phase issues also arising from signal induced and mixed by the separate coils in a fullsize bridge HB? Might cabinet emulation in the circuit be worthwhile? Has anyone tried sending any effects from a DSP unit to the driver?
  18. Pete- I am not thin skinned, so keep it up. I am not trying to be argumentative either, just challenging what may not make sense to me. This is part of the way I learn. I do tend to conced once the understanding arrives. Col- Currently, I am experimenting with cascading 1st and 2nd order all-pass filters, passive and active, after a closely controlled band-pass stage. I am cutting out in my 1st stage anything from below 80 Hz and anything above 2.4 kHz. This is in hopes of reducing the overtones that likely are contributing nothing "good" to the sustain. Results are good so far and I am theorizing this is the key to getting an overall phase correction. I have listened to your clips a few times and they are very nice, my favorite being the atmospheric noodle. Do you still have some fizz ? Listening closely, I can hear some, unless it is my crappy PC speakers. I have a strong fizz now to where distorted playing is my only option, but I have plans on fixing this with novel techniques which redirect EMI to a safer path,rather than attempt to "shield", but it is not my main concern at this point anyway as I am mostly a distortion aficionado. I do understand it affects this channel as well, but I can live with it at this point. I didn't comprehend your statament about broken bends, VST? What is that? Are you sure this is not just the fact that phase shift varies with frequency? What are you basing this assumption on? I can see how something like that might be easily/intuitively supposed, so can you provide a source for the theoretical to educate me? My intuition (though it is just that) would suppose that the macro distance is not the issue as there is a whole number of wavelengths within that distance and as they are whole waves, their effect on shift is inexistent. I would think it is the last, leftover, fractional, tiny, almost quantum portion of the wavelength remainder that causes the shift. Just conjecture at this point, but I want to know more. This is exactly why I am feeling increasingly strong that a high-accuracy reproduction of sound is a necessity of at least equal importance as the driver with regard to the whole system... hence I am growing increasingly anti-LM386, though my latest circuit has finally after hours of tinkering, stabilized it enought to avoid oscillation at full gain. If the latter portion of that is the case, how can anyone (with the currently proclaimed DIY configuration) be getting fundamental or harmonic mode reliably all the way up the neck on all strings? Or are they actually not? How can there be anyone NOT encountering phase problems? Agreed, and I am not sure how I feel about this. I LOVE harmonics. A lot of my lead techniqure revolves around false harmonics. My mixed mode (as I do not yet have a 100% fundamental mode) kills artificial harmonics and brings them to either the fundamental or the harmonic that is dominant at that particular fret/string location. I too share this sentiment, but I do have confidence that much of the work to be done in figuring out the optimal driver has been completed by Pete and I feel confident in believing this and/or at least I feel that the evolution of the circuit is WAY behind that of the driver. I also know my driver is capable of reproducing all required frequenices. My challenge so far has been getting them to all be accesible at the same time. I know little of your driver, but would like to know more, sans trying to find it in this encyclopedia. What I do not feel confident in at all is the whole fetzer ruby idea. It is just not worthy of inclusion in this project IMO and I am continually surprised at complacency with this circuit and that it is not viewed as a huge hindrance to progress. I have looked at your circuit, but do not want to implement anything I don't have a complete understanding of. Being a circuit design noob, I am learning quickly, and the curve is quite steep, which I enjoy, but... I do not have the expertise at this point to look at your schematic and say to myself... "oh yeah, that cap is doing this and this op amp is biased like this to provide xxx" if that makes sense. I would be thankful if at some point you might re-post your latest schematic and give kind of a verbal breakdown of its parts and HOW it functions or rather how its components interact with one another... then again I realize this is a lot to ask of someone. I am also realizing through research that dual power supplies is probably going to be the way to go to eliminate a great deal of these phas issues by removing the need for a great deal of passive components.
  19. I can already reproduce this mechanism, so it is just a matter of packaging it for ease of access. You have mentioned inability to measure/quantify anything, so how do you know this? It actually is fairly simple, regardless of the fact that offset varies with frequency. Implementation is no more complex than using a jfet, opamp, or an LM386. Point me to the points in this thread where others or yourself have tried and failed if this has happened, else I think the assumption can be made that the details of the theory were overlooked or others simply were not interested enough in the idea to pursue it. Again, this is not going to be that difficult to overcome. Humor me for just a moment... if what I am saying is true, wouldn't this be the way to tailor the system to each guitar? [\quote] So again, please point me to the place in this thread where others have tried these things and failed and figured out why. I can without experimentation with this as of yet concede what you say makes sense here and thank you for providing a clearer idea of the speaker functionality. The phase shift occurs within the circuit. This is fairly simple to verify. It doe not occur at the pickup at more than negligible amount. I have measured it several times. The interactions that cause it, yes even the frequency-dependent offset, which are really just filtering interactions, can be neutralized. You have clearly made this point repeatedly. What you have done so far is excellent, and many have enjoyed its fruits. However, not all of us (myself point case) can get the same results as you for whatever the myriad of reasons are. That leaves us wit... the circuit is grossly... well gross in most ways. It does not take much googling on the web to see what a POS chip the LM386 is, honestly it is older than me and so prone to oscillation it's not even funny and there are more folks out there with bad things to say than good. The J201 preamp may be ideal, but I can not prove this as I have little success getting a better result than I can get with a TL082 op amp preamp and this is a mediocre at best op amp. I know the reasons you say LM386 ... it's good "enough" for our purposes, but we don't all have YOUR coils, so the circuit is the ticket and I am surprised you are as complacent with this. There is already the path for simple DIY. Let there be more as well, no need to limit. I did not ignore. Thank you for that. I do feel I understand that better at this point and you are corredt on where I missed this. I was under the assumption the speaker cone was magnetically at rest. So, please confirm so I can be very clear... a speaker cone is not magnetically neutral when at rest? Why did the equal forces stuff you did fail? It seems to me it would be easier to vibrate the strings under tension if they were truly at rest, not being pulled more to one side.
  20. It was a brainstorm for power savings... that was my angle. This is exactly what I meant when I wrote... Kind of like the speaker with an offset in the pull direction, so /push/-/neutral/-/pull/ becomes /pull least/-/pull more/-/pull most/. Basically, pull and different levels of release, but never an actual push to where the string is under zero magnetic influence, whatever that might be. A shame is can't be more like the speaker and have true neutrality. I've heard this many times before and maybe I am misunderstanding, but I often get the notion that you underestimate the driver's capabilities to switch polarity quickly. Of course a halfway decent coil can handle switching states at audible frequencies. If it could not, we would not have speakers capable of high pitches. Even a $2.00 set of headphones can switch within the audible range, so I think it's reasonable to count switching frequency out as a weakness unless the craftsmanship is poor.These cheapo headphones don't seem to have a problem, so what's the difference between them and the driver then? The driver switching speed is not the problem, supplying the signal quickly enough is, but then again not even that is! Supplying a properly phase-corrected signal is. If the signal is properly adjusted, it shouldn't matter if you're 5 or 10 wavelengths behind. I think the biggest problem with this project at least in my case is phase relationship. This can be corrected and I would go as far as to say "corrected" may not even be entirely desirable... given recent experiments, with a little tweaking, I will in one month's time have this "correction" in place and user-adjustable not as a set and forget, but as an on-the-fly, user adjustable phase/fundamental/harmonic control. I have discovered how to tweak my circuit at this point to switch out components which raise the fundamental, the octave harmonic, the 5th harmonic, the and the 3rd harmonic. For the rock player, these Xth non-octave harmonics are much more powerful than the octave harmonic, which suits a more subdued emotion. This past few weeks I have been disappointed, thinking I was getting nowhere, getting killer harmonic sustain (of the Zakk Wylde kind) on some frets and not others, but now I am seeing why and this as an opportunity for improvement and yes I dare say innovation. Think 5-6 distinct modes via selector instead of 2 or even infinite incremental adjustment via knob. I'll be posting clips within the next few weeks.
  21. Hello all- Sorry to hear about the devastation out there, Pete. My sympathies are with you and your people. Let's see what kind of replies this generates. This is regarding the signal sent to the driver. My apologies if this was covered somewhere else in this encylopedia of a thread. Assuming a clean (meaning sinusoidal waveform with no clipping) is being sent to the driver, what is the physics of what is happening to the guitar string? I got to thinking that if the driver were exactly like a speaker (which it is obviously not), then they might exhibit similar behavior... with no signal the cone is at rest, centered between to extremes. Then, with a pure AC signal, there is a pull on the cone during one half (which side is trivial), let's say the positive half of the cycle. Then during the opposite, negative side of thecycle, there is a push, so the cone can move in both positive and negative directions. This bidirectional movement is possible because the cone is centered when at rest. However, our drivers are not centered. I would venture to say they are nowhere near centered for a couple of reasons which I of course am completely, intuitively assuming... 1) The strings are under constant pull force of permanent magnets. 2) The strings, although loosely analagous to the speaker cone... as they are not magnetic, can not ever be pushed, only either pulled or released by the momentary field. 3) The polarity of the permanent magnets employed is never reversed only strengthened and weakened by the driver's induced field. What I mean is that the driver only has the ability to exert different pull values, never a push value. First theory of the day... if number 1 and 2 is correct, then the string need never be pushed. I have a vague feeling though, that one might envision the at rest condition as a pull of say strength 3 (on some made-up scale), then on the half wave peak, pull is up to 5 (strength of +2). Now the wave crosses origin and we're back to strength 3. Once we hit max negative cycle, we can go 2 less than the positive 3 value, bringing it to only a +1, which is essentially less than the at rest state, so both sides of the wave ARE useful. Next theory... if 3 is correct, then there is no sense in trying to push, only pull, and as such half the waveform is useless. Next lame theory... if both 1 and 2 are correct, but lame theory 1 is incorrect, then The only other idea I have... and this one I am not entirely convinced of at this point as it feels far fetched, is that the string is being pulled on both sides of the wave, in effect being pulled at double the audible frequency. To clarify, imagine that the negative sides of the wave were being turned upward, so there were only positive values. What would be the behavior? How would we know if this "driver" was or was not behaving this way? The reason I think it might be is because I was reading about AC electromgnets today and if I understood the material properly, an AC electromagnet with no magnetic core is possible given constanct frequency, meaning at say 60 Hz, the N and S poles change at 60 cycles per second, yet there is a perceived constant pull. There must be more or less a constant pull, as an equal pull then push scenario would lead to net zero pull, no? I would like to hear what you all think is happening on this level. I am making good progress with my units, with more bugs to work out, but at this point, I would really like to grasp this a little better to decide where I am going next. Keep rockin.
  22. Would anyone care to comment on the point I raised previously? One thing I have not seen discussed is the fact that once harmonics bloom, the pickup is then sending this harmonic signal to the circuit, then with circuit amplifying the harmonic, not the fundamental, the driver must be pushing that same harmonic. I wonder what the effect will be if my filter cuts out the higher harmonic frequenices. In other words, if I cut any frequency above a normally fretted 24th fret on high E, any harmonic above that frequency will be ignored, not amplified. I am speculating that assuming there were no phase issues, it might cause an oscillation between fundamental and harmonic, or keep the note just on the edge of becoming a harmonic. What do you guys think?
  23. If you look at the 2nd drawing I posted, I moved C2 (and I think it is called C5 in the second one) right next to the IC chip. Like I said before, there were things about the original F/R layout I didn't like.... The from scratch circuit I am working on will be the 2nd (new version) drawing that I posted. Yes, that ought to keep any noise picked up in between to a minimum.
  24. For fundamental frequencies, I reckon this is the band of frequencies (for a 24 fret neck)... String 6 E = 83Hz open string thru 332hz (highest fret) - however to allow for alternate tunings, say 70Hz thru 400hz String 5 A = 110Hz open string thru 440hz (highest fret) - however to allow for alternate tunings, say 85Hz thru 500hz String 4 D = 146.8Hz open string thru 552Hz (highest fret) - however to allow for alternate tunings, say 110Hz thru 600hz String 3 G = 196.0Hz open string thru 784Hz (highest fret) - however to allow for alternate tunings, say 150Hz thru 820hz String 2 B = 246.9 Hz open string thru 987.6Hz (highest fret) - however to allow for alternate tunings, say 210Hz thru 1050hz String 1 E = 329.6Hz open string thru 1,318Hz (highest fret) - - however to allow for alternate tunings, say 290Hz thru 1480hz but for a sustainer, you'd obviously want the octave harmonic, so when factoring for filters, you'd need to double those figures above. Yes, thank you for the correction! I don't have the foggiest how I ended up with the 8kHz! I am designing an opamp bandpass filter today for the range of 70-2960... will follow with preliminary schematic.
  25. Which capacitor are you refering to? Is it in the part of the circuit that is like the one 1 posted, or is it in your new preamp? I've just gotten the rest of the parts today that I need to build a from-scratch version of my circuit layout (found a few pages back). I know the older F/R-like layout works, but there are things I don't like about it so I'll compare the two, and see if my new layout fixes the problems. I'm also re-wiring the guitar again from the ground up, so hopefully this latest attempt will produce a (consistently) working sustainer. If it does, I'll definitely post some demo sound clips and videos too. I'm going to work on the new unit over the weekend, so we'll see how far I get... Hi MRJ. Good luck with your rewire. Can't wait to hear your sound clips. I am referring to your C2, supply decoupling, bridging Vs and Gnd. From what I have read, that should be placed as close to the IC body as possible as you can pick up noise anywhere in between the cap and the IC and that will help to smooth it out. I experimented with my op amp preamp tonight and found that noise decreases as I raise the value of this cap, but the biggest one I had was 470 uF. A 1000 uF might even be better. One effect of a larger value is when you kill the circuit, it runs for a few extra seconds as the cap discharges. I got my preamp working a lot better tonight. Pete, you are absolutely right about using an AC adapter for testing. It is 100 X noisier than a battery. The one I have is a floating ground. Perhaps one with an earth ground would work. When my band plays, we always use a power conditioner, so I will be curious to see its effects as well. Some interesting effects during testing with a battery... touching nothing but the battery case during play through a speaker increases volume of signal and noise (or gain, not sure), touching nothing but electrolytic capacitor body tops during play quiets noise. These circuits themselves are very prone to picking up noise. One fequency I am picking up is exactly the same frequency as an open B string! Strange. I also found my C7 was an error on my part. I removed it and flipped a couple connections and now 1 less component. Now, it takes the 4.5 VDC and AC swing, compares it to 4.5 straight volts, and amplifies the difference (the AC swing). This is more efficient, BUT, I am beginning to think with these op amps it would be alot better to use 1% tolerance resistors on my voltage dividers (R1 & R2, R7 & R8), as mismatches will lead to amplification of the DC (junk). I also confirmed tonight that the TDA7267 will not accept a volume control pot at all. It must have a capacitor as the last component before the input. I have no idea how it "knows" this. I am going to order some of Strib's IC's (TBA820M) as I think this is a better choice and Strib's sustain sounded nice, even using the antiquated LM741. I am getting nice harmonic sustain, but still no fundamental mode. More testing on the scope is necessary to confirm what I am 99% sure is a phase issue. Next, I am going to create a bandpass filter with a band of 80Hz to 8kHz as this seems to be about the guitar's range and see how this helps my noise issues as I find it more important than missing the fundamental mode. One thing I have not seen discussed is the fact that once harmonics bloom, the pickup is then sending this harmonic signal to the circuit, then with circuit amplifying the harmonic, not the fundamental, the driver must be pushing that same harmonic. I wonder what the effect will be if my filter cuts out the higher harmonic frequenices. In other words, if I cut any frequency above a normally fretted 24th fret on high E, any harmonic above that frequency will be ignored, not amplified. I am speculating that assuming there were no phase issues, it might cause an oscillation between fundamental and harmonic, or keep the note just on the edge of becoming a harmonic. What do you guys think? I do want to get that working, but to be honest, I am in this for the harmonic mode. I can't wait to play Santo and Johnny's "Sleepwalk" live with that! It is going to be so sweet. I also think my driver may be trying to rattle wires inside my guitar! ...or, my driver is slightly microphonic. I can hear a very slight, very high frequency tone emanating from somewhere beneath the pickguard area, but can not confirm the exact source. I also suspect it may be my bridge pickup, Duncan JB full-size. I may rewind it (driver) yet again and try a different potting material. Who knows, Pete, I may even try PVA, but that would be conforming to confirmed methods and we know we can't have that! For anyone interested, here is an awesome link on op amp basics... the whole site in general is an indispensable resource for anyone as new to all this as I am... ALL ABOUT OP AMPS The end of the article makes some recommendations on op amp sourcing by application. There are even op amps that can put out 100's of mA. That might be nice to cut this circuit down to 1 IC. There goes half the components and probably the phase shifts with them! I found some others too that are very fast and clean according to datasheet, much better than TL082 or even 072. Check out OP275. Very low noise, kick-arse slew rate of 22 V/uS which can keep up with changes 44 X the speed of the LM741 and 26 X the speed of the TL082. I think once noise is sorted, this is going to be a better preamp than the fetzer valve. Op amps seem much easier to set gain. You can do it with resistor and pot choices instead of having to bias anything so tediously as the JFET. I think you guys are going for no gain on the JFET, correct? To do this with the opamp, you just leave out the resistors and pot on the negative feedback loop, easy! That is one reason I am staying away from the JFETs. And that reminds me, maybe I need to try no gain. I have been putting in minimums of 2 X.
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