Striker plexi drive schematic

Started by Striker Amplification, July 10, 2013, 06:44:03 PM

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pinkjimiphoton

we love you too, artifus, for sure... you're a very valued and respected member of the community. ;)
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"When the power of love overcomes the love of power the world will know peace."
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armdnrdy

#41
Quote from: pinkjimiphoton on July 12, 2013, 05:16:26 PM
we love you too, artifus, for sure... you're a very valued and respected member of the community. ;)

I believe you hit on a point that a lot of newcomers fail to pick up on.

This is not a blog where people log on to rant/bark about something/anything!

DIYstompboxes is a community with many long term members that should be treated as such. Respect should be shown to those that were here before, because without their loyalty and encouragement, this site may not have survived.

I often find answers to questions in the archives from years ago, written by a member who no longer frequents this site.

One can "register" to access all areas of DIYstompboxes but it is my observation that it takes much more than a simple registration to become a "member."

This site and help from it's members has allowed me to build, modify, and design things that I wouldn't have imagined before!  

I think at the end of the day my best advise is......stick around and assimilate! You just might learn something!
I just designed a new fuzz circuit! It almost sounds a little different than the last fifty fuzz circuits I designed! ;)

Digital Larry

Quote from: armdnrdy on July 12, 2013, 05:45:44 PM
This is not a blog where people log on to rant/bark about something/anything!

You do need to at least consider that Something/Anything is one of Todd Rundgren's most creative endeavors and has some heavy guitar, e.g. "Black Maria".   ;D
Digital Larry
Want to quickly design your own effects patches for the Spin FV-1 DSP chip?
https://github.com/HolyCityAudio/SpinCAD-Designer

artifus

appreciate it guys but i think i was a little out of order - think i was just enjoying myself a little too much and expecting everyone else to know where i was coming from and to share my warped sense of humour. i'm glad it was picked up on. i feel humbled. and thats' no bad thing. peace, y'all. x

armdnrdy

Quote from: Digital Larry on July 12, 2013, 06:00:24 PM
Quote from: armdnrdy on July 12, 2013, 05:45:44 PM
This is not a blog where people log on to rant/bark about something/anything!

You do need to at least consider that Something/Anything is one of Todd Rundgren's most creative endeavors and has some heavy guitar, e.g. "Black Maria".   ;D

I've taken it under consideration and I'll get back to you with a reply. A response to my use of the phrase "Something/Anything" could take some time, possibly as long as a year. (kind of like the government)  ;D
I just designed a new fuzz circuit! It almost sounds a little different than the last fifty fuzz circuits I designed! ;)

Gus

Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?

There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.

http://fuzzcentral.ssguitar.com/schematics/fz1.gif

That one starts off with a no-bias emitter follower!  The transistor starts to conduct when the input voltage swings start to pull some current through the cap.  But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.


That one uses base leakage to get a bias voltage across the base resistor


The circuit here has no resistor at the 2nd devices base so the way it work is the signal from the first stage is cap coupled to the base and if it is large enough one way it turns the device on.

It is two gain stages in series

grounded emitter with a C to B bias feedback resistor is not the most predictable way to bias a circuit

R O Tiree

Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?

There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.

http://fuzzcentral.ssguitar.com/schematics/fz1.gif

That one starts off with a no-bias emitter follower!  The transistor starts to conduct when the input voltage swings start to pull some current through the cap.  But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.


Larry,

That FZ-1 is a positive GND, 3 x PNP circuit, so each stage is biased. This one is negative GND with 2 x NPN and Q2 has no resistors going to its base at all. Therefore, no base current, so Q2C locks up at +9V. In addition, the 1M from Q1C to Q1B is surely providing negative feedback, thus limiting the gain achievable.

I can see how this would work if there was another 1M, for example, from Q2C to Q2B but...
...you fritter and waste the hours in an off-hand way...

earthtonesaudio

Quote from: Gus on July 12, 2013, 06:35:10 PM
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?

There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.

http://fuzzcentral.ssguitar.com/schematics/fz1.gif

That one starts off with a no-bias emitter follower!  The transistor starts to conduct when the input voltage swings start to pull some current through the cap.  But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.


That one uses base leakage to get a bias voltage across the base resistor


The circuit here has no resistor at the 2nd devices base so the way it work is the signal from the first stage is cap coupled to the base and if it is large enough one way it turns the device on.

It is two gain stages in series

grounded emitter with a C to B bias feedback resistor is not the most predictable way to bias a circuit

Quote from: R O Tiree on July 12, 2013, 06:39:57 PM
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?

There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.

http://fuzzcentral.ssguitar.com/schematics/fz1.gif

That one starts off with a no-bias emitter follower!  The transistor starts to conduct when the input voltage swings start to pull some current through the cap.  But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.


Larry,

That FZ-1 is a positive GND, 3 x PNP circuit, so each stage is biased. This one is negative GND with 2 x NPN and Q2 has no resistors going to its base at all. Therefore, no base current, so Q2C locks up at +9V. In addition, the 1M from Q1C to Q1B is surely providing negative feedback, thus limiting the gain achievable.

I can see how this would work if there was another 1M, for example, from Q2C to Q2B but...


Larry's correct and Gus and R.O. are incorrect here.
The Striker circuit, and the FZ-1, bias in roughly the same manner.  That is, leakage between collector and base will cause the transistor to conduct at least partially.  In an NPN it is leakage from collector toward the base (conventional current flow) and with PNP it is leakage from base toward collector, but it's the same thing.  The FZ-1 has some extra resistance in parallel with the base-emitter junction but the principle is exactly the same.

R O Tiree

Got it. Thanks, earthtonesaudio. Still wondering why Ryan didn't answer the question (four times) though...

I gotta cut down on my (ab)use of parentheses as well...
...you fritter and waste the hours in an off-hand way...

Jdansti

Quote from: R O Tiree on July 12, 2013, 08:54:56 PM
Got it. Thanks, earthtonesaudio. Still wondering why Ryan didn't answer the question (four times) though...

Fair enough.  :)

As the last several posts show, there are some folks who understand the way the electrons dance in these circuits (I'm not one of them   :-[).  Ryan is new enough that we don't have a good idea of his understanding of the theory.  There are quite a few of us who come up with circuits without understanding all of the theory behind them.  It could be that he isn't responding to some questions because he doesn't have answers, or it could be that he doesn't like your avatar ;).  Whatever the case, it would help if he responded in some way, and if the answer is "I don't know", that's an acceptable answer.  I can imagine that it's daunting to present your ideas to gurus and try to answer their questions.  There's nothing wrong with gurus asking questions, I'm just saying that some folks may not know how to respond.

Something I've noticed, though, is that there are a bunch of folks who regularly have good, friendly conversations on the forum.  However, there are a few (very few) of the same friendly, helpful people who get snippy or sarcastic very easily (I've never done that  ::)).  And then it doesn't help when the target of the insult fires zingers back (I've never done that either  ::)).  But as Aron said recently, this is supposed to be fun.  There's no need to insult or return insults.

Anyway, I hope that someone gets some benefit from my rant.   :)
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Thecomedian

Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
That's not jimi's point, comedian. His point is that, once you have digitised and compressed an analogue signal to turn in into a lossy mp3, you can never get a truly accurate analogue signal from it. Similarly, if you remove all bar a vanishingly small amount of any part of the frequency spectrum from a signal, you can never get it all back into balance without compromises. Like noise, for example. You alluded to that yourself, when you raised the question of its practicality.

@Ryan - Any insight into how Q2 is being biased? I realise that it's only the third or fourth time that I have mentioned it, but I'm not alone in wanting to know.

oh, well that I would agree with. I didn't catch him talking about lossy digitization. A sine wave has an infinite number of variance, which is a lot more information than just 10 1's and 0's.
If I can solve the problem for someone else, I've learned valuable skill and information that pays me back for helping someone else.

Striker Amplification

It partially has to do with DC current flowing to/in more than 1 area at once. We normally have a bias resistor in place, i have found no need for this because, the current reverse actually takes place within that capacitor, #1: the DC current has already been filtered by our gain stage capacitor, the current flowing out of our cap is now an AC generated SIGNAL. Without the presence of a dominating positive DC voltage, the current gracefully flows to its desired place where needed, if we were talking about a DC signal here then yes a bias resistor would be ideal. DC current flowing from one transistors collector into another transistors base is the perfect example of current shifting, the said DC current is attracted by the opposite in polarity sending said current flowing in the wrong direction. The AC entering Q2's base acts like a completely separate voltage from that of the collector or emitter, it has no polarity, it cannot be redirected by other current flow. there is no current shift because there is no attraction. rendering the bias resistor...........useless. try it. There is absolutely 0% difference with it in the circuit and without it in the circuit.  ;)    
Builds:Plexi superlead mkll 100,plexi superlead mkll 50, 59 plexi, 59 4x10 bassman, marshall bass and PA head 100, JTM45, modded JTM45-lead,JCM800 lead 50.

Lurco


Striker Amplification

Builds:Plexi superlead mkll 100,plexi superlead mkll 50, 59 plexi, 59 4x10 bassman, marshall bass and PA head 100, JTM45, modded JTM45-lead,JCM800 lead 50.

Digital Larry

#54
Kir... Kirrr.... Kirchhofff!!!  

<'scuse me>

http://en.wikipedia.org/wiki/Kirchhoff's_circuit_laws

I'll modify my previous response a bit.  The FZ-1 schematic shows a resistor from base to ground.  If there is DC current flowing from collector to base, and there were no resistor from base to ground, then all collector-base current would be coming out of the emitter.  In the case with a resistor, some of this current will go through that resistor instead and the transistor will be closer to being turned off.  So it's more of an un-bias resistor in the FZ-1's case.

And I see that the AC176 has a nominal spec of Icbo of 35 uA.  

http://pdf1.alldatasheet.net/datasheet-pdf/view/131458/ETC1/AC176.html

And here's a helpful article talking about leakage.

http://www.tpub.com/celec/48.htm

I'll be honest, I learned about transistors in engineering school 30 years ago, but I have reviewed many of my textbooks in the past few years and things which are common practice for stompbox circuits are NOT described, other than as situations to be avoided.  Sending audio through a grounded emitter stage is certainly one of those because of the (naive) assumption that we don't want to distort the signal!   :D :icon_biggrin: ::)  So trying to understand how these circuits work bends my brain in a good way.  I learned something here because I've never done a design where leakage was an important part of the intent.

My only conclusion here is that circuits which depend a lot on the characteristics of the transistor should try to identify what those things are because lots of times, especially with Germanium transistors, somebody's going to try to use a different one.
Digital Larry
Want to quickly design your own effects patches for the Spin FV-1 DSP chip?
https://github.com/HolyCityAudio/SpinCAD-Designer

R O Tiree

Quote from: earthtonesaudio on July 12, 2013, 07:58:32 PMLarry's correct and Gus and R.O. are incorrect here.
The Striker circuit, and the FZ-1, bias in roughly the same manner.  That is, leakage between collector and base will cause the transistor to conduct at least partially.  In an NPN it is leakage from collector toward the base (conventional current flow) and with PNP it is leakage from base toward collector, but it's the same thing.  The FZ-1 has some extra resistance in parallel with the base-emitter junction but the principle is exactly the same.

Having slept on your reply, am I right in saying that, when I suggested a 1M from Q2C to Q2B at Reply#46, that would mimic the leakage from C to B in these old NPN trannies? High-ish leakage, you can get away without it under certain circumstances, low leakage, you'll probably need it?
...you fritter and waste the hours in an off-hand way...

R.G.

Quote from: R O Tiree on July 13, 2013, 05:29:48 AM
Having slept on your reply, am I right in saying that, when I suggested a 1M from Q2C to Q2B at Reply#46, that would mimic the leakage from C to B in these old NPN trannies? High-ish leakage, you can get away without it under certain circumstances, low leakage, you'll probably need it?
Leakage is nearly-constant current source at micro-micro levels. It provides current, but at an incremental impedance that usually way higher than 1M. This was the substance of the Millenium Bypass, and my recently-posted idea to better-fake germaniums by adding a reverse biased germanium diode from collector to base of a silicon transistor to up the leakage to germanium levels.

You may need the current to bias, but its AC impedance is much higher than a resistor.
R.G.

In response to the questions in the forum - PCB Layout for Musical Effects is available from The Book Patch. Search "PCB Layout" and it ought to appear.

pinkjimiphoton

well played ryan!!! :icon_twisted:
make everybody figure it out,  then explain yourself in a way even a dolt like me seems to understand!! :icon_eek:

i'm gonna sit right here....



:icon_mrgreen:

but still... WE NEED VIDEO (or it never happened!!) ;D

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"When the power of love overcomes the love of power the world will know peace."
Slava Ukraini!
"try whacking the bejesus outta it and see if it works again"....
~Jack Darr

Jdansti

>I have reviewed many of my textbooks in the past few years and things which are common practice for stompbox circuits are NOT described, other than as situations to be avoided.

Good point. I've seen R.G. and PRR post similar comments. Many of our stompbox circuits take advantage of distortion by creating "forbidden" circuits.
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R.G. Keene: EXPECT there to be errors, and defeat them...

pinkjimiphoton

Quote from: Thecomedian on July 13, 2013, 03:26:52 AM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
That's not jimi's point, comedian. His point is that, once you have digitised and compressed an analogue signal to turn in into a lossy mp3, you can never get a truly accurate analogue signal from it. Similarly, if you remove all bar a vanishingly small amount of any part of the frequency spectrum from a signal, you can never get it all back into balance without compromises. Like noise, for example. You alluded to that yourself, when you raised the question of its practicality.

@Ryan - Any insight into how Q2 is being biased? I realise that it's only the third or fourth time that I have mentioned it, but I'm not alone in wanting to know.

oh, well that I would agree with. I didn't catch him talking about lossy digitization. A sine wave has an infinite number of variance, which is a lot more information than just 10 1's and 0's.

i was talking about throwing away frequencies in general. it would seem to my uneducated posterior, in theory and in limited practice, that once you attenuate frequencies to ground potential, they, and the area they once occupied would be gone, even from a sine wave.  if you're merely "turning them down" with an adjustable device on a recorded (not live) media, yah, turn it back up and no harm no foul.

but in this case, the shape of the sine wave would seem to be altered permanently.... you would need something to restore them to reconstruct the original sine wave, right? "de-amplifying" on a LIVE instrument tho, i think may be different. with recorded media, yah, the original sine wave is gonna remain intact no matter how much you molest it with filtering or amplification or modulation. it can be restored immediately and at any time.

but wouldn't a live signal, being transient, be forever altered? i mean, obviously you can play the same thing the same way thru the same stuff again to get there, i guess... but once something is gone, to me, it would seem to be really hard to replace later. i mean, how can you boost stuff that you've already cut? wouldn't it be better not to cut it in the first place?

man i'm tieing myself into a granny here, i know it..

i've found sometimes that larger freekin' output caps can sound great. but i would think whatever attenuation occurred prior in the circuit could only be amplified...not restored.

so whatever shape or *distortion* of the original sine had been twisted into can be made louder.. but is it actually replacing the signal lost by storing the charge in the larger capacitor? or just amplifying the curve of the audio signal more?

i guess maybe i should fire up my venerable heathkit scope (my ancient tektronics one hit the floor and died...sad tragic story, snif..) and look..

sorry to hijack this thread, i don't think my understanding is up to snuff..

but yes, i was talking like mp3, where loss is VERY noticeable and pretty much impossible to repair.
  • SUPPORTER
"When the power of love overcomes the love of power the world will know peace."
Slava Ukraini!
"try whacking the bejesus outta it and see if it works again"....
~Jack Darr