Speaker Resonance Controls

Thanks for this information!!! Truly valuable! Amazing results!

I have been studying amp design recently, as I realized my knowledge on the subject was minimal. Although I still have only a tiny bit of knowledge on the subject, I find that it helps me understand posts like these.

This control has a profound effect on the speaker-amp combination on many levels.

Thanks for the info, and such an amazing machine!

The inclusion of the Speaker Resonance and Transformer Match parameters opens up an whole new array of possibilities. Between the two you can get any combination of preamp vs. power amp distortion you want, and mold the character or voicing of the power amp distortion, something you can't even do with a real amp. Very cool stuff.
 
I've been studying the power amp behavior in great detail. I'm not complaining, it sounds good, and it does what it needs to do with two exceptions. The main issue I see is that only the primary impedance of the output transformer changes as the XFormer Match parameter is varied. The other less serious issue is the lack of any change in the sound as the XFormer Drive is varied, at least not to my ears.

With no change in the secondary impedance of the OT the XFormer Match has no effect on the frequency response of the amp given a fixed speaker impedance curve. Therefore it does nothing but create uniform frequency independent gain in the power tubes. I hear no difference between increasing the XFormer Match vs. increasing the master volume control and decreasing the overall level to compensate for the volume and power increase when doing so. Is there a difference between increasing the primary impedance of the OT to create distortion and attempting to keep the current constant vs. increasing the current and keeping the primary impedance constant? I just don't hear any difference.

In essence, it seems to me that increasing the XFormer Match does the same thing as turning up the master volume and reducing the overall level to compensate for the volume increase. Is this how the XFormer Match is modeled? Or am I not hearing the difference? In one case the power tubes are drawing a lot more current to produce the distortion and in the other case (increasing the XFormer Match) the power tubes carry much less current and distort due the to a voltage rise in an increasing impedance.

Both methods can be used to cause the speaker impedance to create frequency dependent distortion with the XFormer Match having a greater range of boost/cut, but both methods seem to produce the same result, with the master gain/level control method varying the output power by increasing the current while the XFormer Match should actually reduce the power and create distortion by increasing the primary impedance but at the same time lowering the secondary impedance if this were a true model of a transformer.

The bottom line is it appears to me that the Xfromer Match isn't actually changing the ratio of the OT, but instead "simulating this effect" by increasing the gain (and current) in the power amp and simultaneously reducing the overall level parameter to change the power gain into an apparent power reduction. An active tone stack placed in the same place as the master volume (post preamp / pre-phase inverter) will act just like the speaker impedance curve. The tone stack and the speaker impedance curve both behave as simple equalizers and the Xformer Match will work the same with a tone stack EQ curve as it does with the speaker impedance curve. In fact the two can be used together to create a more radical speaker EQ curve.

The only effect in the end is what frequencies cause power amp distortion. The fact the ratio change of the OT has no effect on the frequency response of the speaker is a contradiction that implies that the XFromer Match parameter is a "simulation" and not a real circuit. Why does the secondary impedance remain constant?

That's just what I'm hearing. I could be wrong. I would have to run tests with a signal generator and a scope to verify my theory. Perhaps the TigerSharc is not powerful enough to model a real transformer or it just hasn't been perfected and implemented yet.

Again, I could be wrong, and I'm not criticizing, I'm just making an observation. I know a lot of this will be remodeled in 6.0. The only reason I'm getting so deeply into this is that I'm afraid to write patches in 5.07 that will probably be useless in 6.00 and I'm waiting for a set of new Seymour Duncan pups for my guitar before I start permanent patch composition.

I don't have a theory as to why I can't hear the XFromer Drive parameter.

And please don't take this as criticism, and if I'm wrong about any of all of this I'd appreciate being set straight.

Thanks.
 
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There's one more thing that confuses me. If the secondary impedance remains the same at any transformer turns ratio (as evidenced by the lack in change of frequency response as the ratio is changed), doesn't this require that the output tubes change their output impedance radically? Wouldn't the output impedance of the power tubes have to increase radically to keep the secondary impedance from decreasing as the transformer turns ratio is increased (according to the square of the turns ratio) and wouldn't the output impedance of the power tubes have to drop dramatically to keep the secondary impedance from increasing as the turns ratio is decreased?

A real transformer reflects impedance both ways, not just from one side to the other,

IMHO the output transformer is the most difficult part to model in an amplifier. In all the PSPICE amp simulations I've ever written I've "cheated" on the OT. However, these simulations use a fixed ratio simplified transformer "model". Changing the ratio of the OT is a complicated affair, and I can envision some of the "problems" that a true simulation of the range of ratios that the XFormer Match control allows would create. For one thing the impedance of the secondary would change so radically that the effect of speaker impedance curve would end up causing the resultant frequency response to have peaks and valleys in excess of what the hardware could effectively simulate and it probably wouldn't sound very good. I'm just curious why the relationship Zp = N^2 * Zs was cited (in the wiki) when Zs seems to remain fixed. If Zs is fixed and N is radically changed, then Zp must radically change. But Zp is dependent on the output impedance of the power tubes. Therefore shouldn't they be forced to change their impedance as well, and radically so? That simply doesn't seem indicative of the way real amps and transformers function.

Again, not criticism, just confusion about how this is all actually modeled. If anyone can shed some light on how a power tube can radically change its impedance or where my logic is flawed I'd appreciate it. If you don't care I can understand that too!

Thanks.
 
There's one more thing that confuses me. If the secondary impedance remains the same at any transformer turns ratio (as evidenced by the lack in change of frequency response as the ratio is changed), doesn't this require that the output tubes change their output impedance radically? Wouldn't the output impedance of the power tubes have to increase radically to keep the secondary impedance from decreasing as the transformer turns ratio is increased (according to the square of the turns ratio) and wouldn't the output impedance of the power tubes have to drop dramatically to keep the secondary impedance from increasing as the turns ratio is decreased?

A real transformer reflects impedance both ways, not just from one side to the other,

IMHO the output transformer is the most difficult part to model in an amplifier. In all the PSPICE amp simulations I've ever written I've "cheated" on the OT. However, these simulations use a fixed ratio simplified transformer "model". Changing the ratio of the OT is a complicated affair, and I can envision some of the "problems" that a true simulation of the range of ratios that the XFormer Match control allows would create. For one thing the impedance of the secondary would change so radically that the effect of speaker impedance curve would end up causing the resultant frequency response to have peaks and valleys in excess of what the hardware could effectively simulate and it probably wouldn't sound very good. I'm just curious why the relationship Zp = N^2 * Zs was cited (in the wiki) when Zs seems to remain fixed. If Zs is fixed and N is radically changed, then Zp must radically change. But Zp is dependent on the output impedance of the power tubes. Therefore shouldn't they be forced to change their impedance as well, and radically so? That simply doesn't seem indicative of the way real amps and transformers function.

Again, not criticism, just confusion about how this is all actually modeled. If anyone can shed some light on how a power tube can radically change its impedance or where my logic is flawed I'd appreciate it. If you don't care I can understand that too!

Thanks.

I would really like to understand what youre talking about :D but thanks to guys like you we have got all the musical nuances :))
 
I would really like to understand what youre talking about :D but thanks to guys like you we have got all the musical nuances :))

Sorry for the long posts, and I could be wrong about some of this, but I still hear the same things I described. I don't know if any of my posts will make any difference in the future firmware, but I feel compelled to share my perceptions and hope to get the perceptions of others and maybe a patch that clearly demonstrates I'm wrong. I have no problem being wrong.

Until FW 6.00 comes out all of my observations are moot to an extent, but if it helps provoke an improvement in future FW even if it's 7.0 or 8.0 then it was worth it.

If I had my druthers I would reduce the range on the Xformer Match control.

0.1 to 10.0 is a range of 100, and if that represents the turns ratio then it represents an impedance range of 10,000:1! It doesn't sound to my ears like it's changing the impedance ratio by a factor of 10,000. I'm guessing the parameter values must represent the impedance ratio and not the turns ratio based on what I'm hearing. Even an impedance ratio of 100:1 (the range of the values of the control) is a huge range.

Assuming it DOES represent the turns ratio, if the range were changed to values between 0.5 to 2 with 1.0 being the nominal ratio, that would represent a change in the turns ratio of 4, and an impedance range of 16:1 instead of 10,000:1. Changing the impedance ratio of the OT by a factor of 16 is still huge and would have a dramatic effect on the sound in a real amp. I just don't get the 0.1 to 10 range - it seems like something that would be impossible to model and have work properly. If the delta in the turns ratio were limited to 4:1 it seems to me that a realistic model could be developed that reflected the impedance of the tubes back to the secondary which drives the speaker and there could be realistic interaction between the output impedance of the secondary and the impedance curve of the speaker.

As it stands now, it functions, and having the control is infinitely better than not having the control, even if it doesn't behave like a real amp. But the distortion produced by pushing too much current through the power tubes into a low impedance primary of an OT is not the same as the distortion produced by a low power tube current into a high impedance OT to create voltage clipping since the voltage is limited by B+, and these two different ways of creating distortion operate on a different part of the non-linear curve of the power tubes in a real amp, but to my ears the XFormer Match sounds the same as whatever the master volume control is doing which in a real amp is increasing the current in the power tubes into a fixed impedance.

Again, not a complaint, just an observation.

To go off topic for a moment, I've been studying the Analog Devices website and reading their introductory literature, and also started learning C++. Analog Devices has something called DSP++ that runs on a PC and can be used to simulate and develop DSP applications that will run on their products including the TigerSharc. All I can say is this stuff gets complicated!!! DSP++ is also extremely expensive although you can get a 90 day evaluation for free. I'm not even close to being able to use it though.

All the theory associated with this picks up just a little bit past where I left off in school, and I think it will take years to get to a point where I could create any sort of useful design of even a simple device. But you have to start somewhere. I was taught Fortran, and I was quite good at it, but it's a dinosaur and useless for DSP. Just learning C++ at an intermediate level is a challenge.

Anyway, the suggestion to change the range of the Xformer Match should be taken with my humility in not understanding the process by which this is all being modeled. I'm not trying to be a know it all because I'm far from it by my own recognition and admission. I wish I knew more about this stuff but will be slowly acquiring knowledge pertaining to how DSP works. I have a lot of high level math to learn (and some to relearn) while I'm at it. I'm giving myself about 5 years to come up to speed, longer if necessary. I'm an analog circuit designer, always have been, and now I want to learn how to simulate the stuff I've designed and sometimes built using DSP instead of etching custom boards and soldering analog parts onto them.

I have one more thought that I have to spit out. I was wondering if it would be worth it to create a low res amp block model that used all the power of one processor. This would (theoretically I think) allow twice the complexity of the circuit model used in hi res mode and perhaps allow a complete component level model of an amp with no shortcuts or simulations of circuit behavior. The model would work exactly like a real tube amp since it would have nothing but discrete circuit elements. Just a thought, again submitted with the humility that I have no idea what this would entail or the degree to which the existing hi res amps are modeled at the discrete component level.

The fact that a speaker with an impedance that varies greatly over it's frequency range doesn't have a changing frequency response when the turns ratio of the output transformer is radically altered is proof beyond any doubt that the power amp's output transformer or adjacent components are not discrete circuit models but rather artificial simulations. There's nothing wrong with that other than the fact the controls (which in all fairness don't exist on real amps) don't behave like they would if they actually did exist on real amps.

Thanks.
 
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These controls are so powerful that I made a separate GUI page and graph for them. Don't overlook their importance in your tone-shaping quests. I always start with these before resorting to the graphic EQ or EQ blocks (in fact I almost never use any EQ). If you have some favorite speakers you can try to find published impedance data to help match the response.

I just experimented with these controls for the first time and they are an amazing tool for helping me get the sounds I am looking for. Thanks!

I had not been able to get my Mesa Boogie 50/50 power amp going into two Bagend 1x12 cabinets to sound good at all with the Axe FXII. It sounded terrible. I finally got everything to sound good by leaving the amps on in global, but turning off the cabs, and turning the sag off completely in the dynamics section of the amp block. Learning how important the Master volume control in the amp block was to the sound was really important, and the Graphic EQ in the amp block is really useful. By tweaking the speaker resonance controls in the amp block I was able to make my 1x12 cabinets sound more like my 4x12 Marshall greenback cabinets. So thanks to everyone on the forum sharing their knowledge, I am finally figuring out just what the Axe FXII is really capable of. I can't wait for 6.0.
 
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This is a bit old thread but really important! I wish this subject could be explained thoroughly and some examples of real tweaks added here. I play through a pair of QSC K12s / directly to FOH. Now the question is... Should I try to match SPKR parameters with used IR or with my monitors? It's possible to add impedance and eq-curves to IRs, has somebody tried those, any good?
 
This is a bit old thread but really important! I wish this subject could be explained thoroughly and some examples of real tweaks added here. I play through a pair of QSC K12s / directly to FOH. Now the question is... Should I try to match SPKR parameters with used IR or with my monitors? It's possible to add impedance and eq-curves to IRs, has somebody tried those, any good?

It should be matched to the IR, not the monitors. The effect cannot be baked in to the IRs, as it is affecting power amp behaviour and feel, not just frequency response.
 
Thanks antcarrier! I started to read the theory around the topic... The resonance peak occurs around 110Hz and corresponds with a cab size. A "naked" speaker's peak is lower. If an IR has an build-in impedance curve, 50% wet means about 3dB cut to the mids, or a corresponding 3dB boost at the resonance point. The upper-mid rise is a quite difficult to reproduce using EQ. In loud volumes the SPKR FR-curve is about similar to its impedance curve. Etc etc... There is a heck of a lot going on that, at least I, don't really wouldn't mind to understand precisely. I just like to have the best available 90 db'ish live monitor/FOH sound immediately. It would be so much more fun to play than tweak hours and hours. So, all of you who really understand what to do, please write theory and instructions how to really tame the this beast :evil I would so much appreciate it, thanks!!!
 
This page here is a good place to start looking: Scott Peterson: tips and tricks - Axe-Fx II Wiki It has some great tables that are a good reference for setting lo res for different cabs.
There has been some debate regarding whether or not hi freq and mid res should be adjusted (IMO just try it and go by ear!) - also one should note that the mid res section is to be removed in the upcoming V10, so may not be worth the effort in the long-term.
Good luck!
 
Bumping this, partly because it's how I got to finding tones that convinced me to keep the Axe.

These controls are so powerful that I made a separate GUI page and graph for them. Don't overlook their importance in your tone-shaping quests. I always start with these before resorting to the graphic EQ or EQ blocks (in fact I almost never use any EQ).
 
This is a bit old thread but really important! I wish this subject could be explained thoroughly and some examples of real tweaks added here. I play through a pair of QSC K12s / directly to FOH. Now the question is... Should I try to match SPKR parameters with used IR or with my monitors? It's possible to add impedance and eq-curves to IRs, has somebody tried those, any good?

Yes, i´ve tried it !
I have a program called mir/ir made by redwires.
If you look at the imp curves (of a certain speaker modell) they are more exact and got more character than just a bass resonance and a slight treble rise. They have different aeras where the dip and peak and and might be part of the actuall speakers sound. I also think they even have different imp-phase/frequenzy than the one in the axe.
But I don´t know for sure. It doesn´t hurt trying it anyway. Hovever, I´m not sure it makes the sound much better in the end.
I have noticed that if you add a bit of g12m imp curve to an IR there comes a little familiar color out of it that some of us might like.
Also it might be worth concidering to decrease the resonaces in the axe if you mix in these other curves in the IR.

I believe that the sound of the impedance curves in the axe (and an amp) changes when the power amp distorts.
So it might be wrong mixing in a static "curve".

On the other hand , it won´t hurt you.
 
Thanks Thomas and others! I learned that QSC K12s are not so flat at all QSC Audio Forum • View topic - Frequency Response Charts and made about matching corrections to global eq with great success! Of course I turned the lowest and highest totally down as they do not fit in normal speaker response curve. Trying to fix the boom with GEG and PEG was not so good approach, too much of them would have been needed. The other crucial finding is to add a MBC block last in chain and set low/high frequencies about 118 and 4,000 Hz, and tweak their parameters a bit. That helped me to get rid of sounding boomy with different volumes. I have tried other things as well like messing with IRs and speaker resonance controls but as they are meant to be as they are originally set, I can now leave them as is. The conclusion is that what ever FRFR monitors are used, they are just, er... monitors...
 
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