The next thing I noticed is how much distortion in the model the output filter causes. But, when one adds some semblance of a load, without it you get instability and other such bad things. This leads me to a couple of conclusions:
The amp and the speaker must be designed together as a system. Output filtering and compensation taken as one.
Moving to active crossovers to eliminate the inductors in the crossover may do more for the result than this week's fancy VAS.
What I don't know: How bad is the distortion in an active crossover?
This is one of those things when you look at it you have to wonder how it works at all. My hat's off to those who have actually made it pretty darn good.
The amp and the speaker must be designed together as a system. Output filtering and compensation taken as one.
Moving to active crossovers to eliminate the inductors in the crossover may do more for the result than this week's fancy VAS.
What I don't know: How bad is the distortion in an active crossover?
This is one of those things when you look at it you have to wonder how it works at all. My hat's off to those who have actually made it pretty darn good.
That makes me think, the output stage is (depending on design, but is very often) essentially a buffer, a current gain stage. Why not take the global feedback point from the INPUT of the output stage, instead of the actual output stage? This would surely isolate the feedback path from any variations in speaker impedance.
On the other hand, there's got to be something "wrong" with the idea, because I've never seen it done.
On the other hand, there's got to be something "wrong" with the idea, because I've never seen it done.
Nothing is wrong with this idea. And it sounds much better than with single global feedback loop.
Well, may be "wrong" is the result of measurements that designers and manufacturers want to show in order to get proper sales figures. But boutique designers don't need that, so they use it quite often without any odd noise. Except "No global feedback", or "low global feedback" phrase. That means "high" feedback may exist, but outside of "global" loop.
Make the active crossover a digital one (and FIR to bypass any possibility of limit cycles and hence noise modulation, or use double precision IIR) and the distortion's just that of the DAC. Pick your DAC poison carefully
Hi,
There are many forms of mains transformers. The eastern block was big on double-C-Core and LL.
These are in my limited experience (I only tested a few samples of each type) the best choice for audio mains transformers, if correctly wound. Best in fact are 400Hz double-C-cores (made for Ships, Planes, military etc.) which need to be de-rated for 50Hz service, with a tiny airgap.
These are then wound using the knowledge of how audio transformers are wound to generate transformers with a defined, narrow passband (including some passives build into the transformer) and very large common mode rejection ratio all the way to very high frequencies.
It does cost more, but if you buy 100's or 1,000's of pieces it is normally a small price increase and no problem, though the transformer makers will think you crazy applying so much effort for mains transformers, they all are happy to take your money.
Torroids are by far the worst solution to audio mains transformers, though if you know and understand what their problems are you may be able to compensate for them using external means and at least mitigate some of the issues. Again, I see this extremely rarely implemented in any audio gear and given the preponderance of badly implemented torrid torroids in audio gear it is little surprise that the mains filter industry is booming.
In fact, the whole mains powering implemented without thorough understanding of what goes on (and hence creating problems) accounts probably for 70% or more of the sonic differences reported for interconnects and mains cables as well as other mains related items and surprisingly it remains "unmeasurable" as the meter readers often are as clueless as the high end designers of what is going and in absence of measured differences and the presence of claimed sonic differences you get the usual fight...
To a real pro neither fact is surprising and they are trivial to reconcile, so we usually look 
at the hissy fits and general spats that break out as soon as someone mentions sonic differences caused by cables and wire.
Ciao T
There are many forms of mains transformers. The eastern block was big on double-C-Core and LL.
These are in my limited experience (I only tested a few samples of each type) the best choice for audio mains transformers, if correctly wound. Best in fact are 400Hz double-C-cores (made for Ships, Planes, military etc.) which need to be de-rated for 50Hz service, with a tiny airgap.
These are then wound using the knowledge of how audio transformers are wound to generate transformers with a defined, narrow passband (including some passives build into the transformer) and very large common mode rejection ratio all the way to very high frequencies.
It does cost more, but if you buy 100's or 1,000's of pieces it is normally a small price increase and no problem, though the transformer makers will think you crazy applying so much effort for mains transformers, they all are happy to take your money.
Torroids are by far the worst solution to audio mains transformers, though if you know and understand what their problems are you may be able to compensate for them using external means and at least mitigate some of the issues. Again, I see this extremely rarely implemented in any audio gear and given the preponderance of badly implemented torrid torroids in audio gear it is little surprise that the mains filter industry is booming.
In fact, the whole mains powering implemented without thorough understanding of what goes on (and hence creating problems) accounts probably for 70% or more of the sonic differences reported for interconnects and mains cables as well as other mains related items and surprisingly it remains "unmeasurable" as the meter readers often are as clueless as the high end designers of what is going and in absence of measured differences and the presence of claimed sonic differences you get the usual fight...
To a real pro neither fact is surprising and they are trivial to reconcile, so we usually look
at the hissy fits and general spats that break out as soon as someone mentions sonic differences caused by cables and wire.Ciao T
Hi,
Without seeing the Schematics I cannot comment how to change one into the other. What does interest me though, which one is it that your wife prefers? Am I correct that it is the second one?
Then something is not right. The output filter should not add distortion. If it does, it means you introduced an unfortunate interaction between it and the amplifier and load.
It helps, but rarely feasible.
So much should be obvious from at the stability requirements of feedback amp's.
To make this more accessible conceptually, IMNSHO the compensation should make the amplifier stable with resistive loads, the build-out network should resist RF ingress and decouple the Amplifier sufficiently from the load to ensure unconditional stability as system.
Maybe, maybe not.
Invariably worse than that of a passive one. That much should be abvious, assuming quality components are used for the passive crossover (so not ferrite cored chokes and bipolar electrolytics).
Sure, but as in a well designed amplifier the output stage should be the majority contributor to non-linearity (if it is not, linearise the other stages first by using degeneration or local loops and make them more linear), why not drop feedback alltogether (hint - because our output is in class AB, that is why we cannot drop looped feedback - make a true class A Amp and run feedbackless is trivial).
It actually has been done. Krell comes to mind, they had a range of amplifiers with dynamic biasing that kept the output stage in Class A and no looped feedback from the output stage (but local loops from the VAS out).
The Cello Encore Amplifier (and AFAICT others from Cello) used no looped NFB but a variant of hawkesford error correction (which actually is fundamentally again NFB) at low frequencies and a complex set of networks to add some feedback at higher frequencies, keeping Class AB distortion contained.
In the DIY scene the LC Audio Zapsolute and Millenium are examples of Amplifiers that use only local degeneration and no looped feedback, commercially the Etalon Amplifiers from Hungary come to mind.
Former DIYA contributor "Jocko Homo" used to remove the global feedback loop to the input stage from his amp's and instead used a feedback loop around output stage and VAS (which was supposedly excellent sounding) and got promptly crucified by some lippy sh...t at the JAES when he tried to share some of those ideas...
There are other topologies that have been running without looped feedback too. For a really freaky one look up S.P.L.I.F. (I need to that Charles is a sly old Fox - if he was American he would named his idea J.O.I.N.T.).
Generally the price you pay for these topologies is greater measured THD, so Members of the holy Catholic Church of the Meter Readers and Double blinders (which are out in force at thishere Venue and show little respect for differing views) will promptly declare you a Heretic and set the Holy Inquisition on you should you be found to promote such circuitry...
As an aside, these low or no feedback around the output stage topologies usually benefit from "overbias", as this reduces the higher order components, so the distortion result is more akin to a good tube Amp (note, it does not make an SS Amp sound like a tube Amp)...
Ciao T
Without seeing the Schematics I cannot comment how to change one into the other. What does interest me though, which one is it that your wife prefers? Am I correct that it is the second one?
Then something is not right. The output filter should not add distortion. If it does, it means you introduced an unfortunate interaction between it and the amplifier and load.
It helps, but rarely feasible.
So much should be obvious from at the stability requirements of feedback amp's.
To make this more accessible conceptually, IMNSHO the compensation should make the amplifier stable with resistive loads, the build-out network should resist RF ingress and decouple the Amplifier sufficiently from the load to ensure unconditional stability as system.
Maybe, maybe not.
Invariably worse than that of a passive one. That much should be abvious, assuming quality components are used for the passive crossover (so not ferrite cored chokes and bipolar electrolytics).
Sure, but as in a well designed amplifier the output stage should be the majority contributor to non-linearity (if it is not, linearise the other stages first by using degeneration or local loops and make them more linear), why not drop feedback alltogether (hint - because our output is in class AB, that is why we cannot drop looped feedback - make a true class A Amp and run feedbackless is trivial).
It actually has been done. Krell comes to mind, they had a range of amplifiers with dynamic biasing that kept the output stage in Class A and no looped feedback from the output stage (but local loops from the VAS out).
The Cello Encore Amplifier (and AFAICT others from Cello) used no looped NFB but a variant of hawkesford error correction (which actually is fundamentally again NFB) at low frequencies and a complex set of networks to add some feedback at higher frequencies, keeping Class AB distortion contained.
In the DIY scene the LC Audio Zapsolute and Millenium are examples of Amplifiers that use only local degeneration and no looped feedback, commercially the Etalon Amplifiers from Hungary come to mind.
Former DIYA contributor "Jocko Homo" used to remove the global feedback loop to the input stage from his amp's and instead used a feedback loop around output stage and VAS (which was supposedly excellent sounding) and got promptly crucified by some lippy sh...t at the JAES when he tried to share some of those ideas...
There are other topologies that have been running without looped feedback too. For a really freaky one look up S.P.L.I.F. (I need to that Charles is a sly old Fox - if he was American he would named his idea J.O.I.N.T.).
Generally the price you pay for these topologies is greater measured THD, so Members of the holy Catholic Church of the Meter Readers and Double blinders (which are out in force at thishere Venue and show little respect for differing views) will promptly declare you a Heretic and set the Holy Inquisition on you should you be found to promote such circuitry...
As an aside, these low or no feedback around the output stage topologies usually benefit from "overbias", as this reduces the higher order components, so the distortion result is more akin to a good tube Amp (note, it does not make an SS Amp sound like a tube Amp)...
Ciao T
Hi,
In my limited Experience FIR filters are inimical to perceived "good sound". Correctly done IIR filters behave the same as classic passive/analog ones.
So the only benefit of digital crossovers is ease of adjustment including compensation for non-ideal drive units in speakers.
This is what argues strongly in favour of such units for DIY, if the digital crossover can be of sufficiently high quality, but less so for commercial designs where the resources exist to optimise passive system.
Ciao T
Make the active crossover a digital one (and FIR to bypass any possibility of limit cycles and hence noise modulation, or use double precision IIR) and the distortion's just that of the DAC. Pick your DAC poison carefully
In my limited Experience FIR filters are inimical to perceived "good sound". Correctly done IIR filters behave the same as classic passive/analog ones.
So the only benefit of digital crossovers is ease of adjustment including compensation for non-ideal drive units in speakers.
This is what argues strongly in favour of such units for DIY, if the digital crossover can be of sufficiently high quality, but less so for commercial designs where the resources exist to optimise passive system.
Ciao T
Perhaps the FIR filters you've experienced haven't been 'correctly done' ? Care to share more details?
You use 'so' but I can't see how this (above) statement follows reasonably from your earlier ones.
So then air-cored chokes? You don't count non-linear frequency dependent losses as distortion then?
Hi,
Could be they are not "correctly done", though fundamentally a FIR seems to designed "contra lex naturae". As to examples, 99% of all digital audio products. If they create such mayham as they do at frequencies at the edge or beyond hearing, what will they do in the Audio Band.
All digital XO's I have so far encountered where thankfully IIR. The old Yamaha one with PCM63 in the output's is worth hunting down and refurbishing, it is quite horrorshow and will knock any Behringer into a crocked hat. Sadly they seem mostly be bought for salvage, so they are rare.
If the digital XO does not deliver performance that is superior to passive crossovers (and it does not), the main reason one may wish to still employ one is because it is easy to adjust, unlike a passive, analogue crossover.
Hence my "So"...
Ciao T
Could be they are not "correctly done", though fundamentally a FIR seems to designed "contra lex naturae". As to examples, 99% of all digital audio products. If they create such mayham as they do at frequencies at the edge or beyond hearing, what will they do in the Audio Band.
All digital XO's I have so far encountered where thankfully IIR. The old Yamaha one with PCM63 in the output's is worth hunting down and refurbishing, it is quite horrorshow and will knock any Behringer into a crocked hat. Sadly they seem mostly be bought for salvage, so they are rare.
If the digital XO does not deliver performance that is superior to passive crossovers (and it does not), the main reason one may wish to still employ one is because it is easy to adjust, unlike a passive, analogue crossover.
Hence my "So"...
Ciao T
Curious - to me the IIR is like that and the FIR not.
Well, agreed that at least 99% of implementations aren't particularly good ones - half-band and linear phase, equiripple filters abound.
I believe IIR is equally capable of that. You're speaking of ringing? Or only pre-rnging?
So you've never encountered a Behringer? They most certainly have FIR inside the DACs they use (AKM is it?).
TVR was asking originally about distorton performance, you broadened the discussion to sound quality. So by 'performance' here which are you referring to?
Ah now you've qualified the 'so' with 'mainly' that's a material change.
Hi,
FIR is zero/linear phase and so not found in nature, hence my "contra lex naturae" quip, IIR is minimum phase and hence mirrors nature.
I have used the DEQ, mine had CS ADC/DAC.
Depends on the definition of "distortion".
To me, if a deivce/process etc. alters the sound so that what appeared "good" before and "bad" after, it adds distortion, not necessarily measurable in the THD&N domain, but distortion nevertheless...
Not really, except for hair-splitters. Prior I qualified with "seems"...
And I retain, I see no benefit to digital XO's (over analog solutions), except easy adjustability.
Ciao T
FIR is zero/linear phase and so not found in nature, hence my "contra lex naturae" quip, IIR is minimum phase and hence mirrors nature.
I have used the DEQ, mine had CS ADC/DAC.
Depends on the definition of "distortion".
To me, if a deivce/process etc. alters the sound so that what appeared "good" before and "bad" after, it adds distortion, not necessarily measurable in the THD&N domain, but distortion nevertheless...
Not really, except for hair-splitters. Prior I qualified with "seems"...
And I retain, I see no benefit to digital XO's (over analog solutions), except easy adjustability.
Ciao T
FIR is no more linear phase than red wine is Bull's Blood. PCM5102 has a min-phase FIR option.
I don't mind at all if you're not looking for all the other reasons
Regarding having or not having overall feedback, in my experience, I have never heard a device without a global feedback loop which sounded right to my ears. They do tend to have a warmis sound, it tends to be sweet and often a little too sweet, and they invariably leave me with a feeling of a job not finished, as if something was left over. Multiply everything said above by two if it uses tubes.
Which is why I don't much like tube audio, but hey - live and let live.
That said, in my experience, the quality of sound does not depend much on the amount of feedback, but does depend in my view heavily on how that feedback is applied. In this respect, I am totally with Thorsten, I also believe (based on personal experience) that devices with a lot of local but low overall NFB tend to sound better to much better than devices with low local and heavy global NFB.
The fact that those with heavy global NFB ALWAYS have better published specs means absolutely nothing to me (although it's easy to see why they mean so much to commercial manufacturers, people actually do read spec sheets, in fact they rely on them too much and certainly much more than on their own ears), I honestly don't give a damn if their published specs say 0.001% or 0.3% THD if it sounds good to me.
A great example is my own Harman/Kardon 6550 integrated. It's rated at 0.3% THD into 4 Ohms, 20-20,000 Hz at rated power, but it actually sounds a hell of a lot better than most other amps with specs like 0.01% under the same conditions. It uses just 17 dB of global NFB.
Which is why I don't much like tube audio, but hey - live and let live.
That said, in my experience, the quality of sound does not depend much on the amount of feedback, but does depend in my view heavily on how that feedback is applied. In this respect, I am totally with Thorsten, I also believe (based on personal experience) that devices with a lot of local but low overall NFB tend to sound better to much better than devices with low local and heavy global NFB.
The fact that those with heavy global NFB ALWAYS have better published specs means absolutely nothing to me (although it's easy to see why they mean so much to commercial manufacturers, people actually do read spec sheets, in fact they rely on them too much and certainly much more than on their own ears), I honestly don't give a damn if their published specs say 0.001% or 0.3% THD if it sounds good to me.
A great example is my own Harman/Kardon 6550 integrated. It's rated at 0.3% THD into 4 Ohms, 20-20,000 Hz at rated power, but it actually sounds a hell of a lot better than most other amps with specs like 0.01% under the same conditions. It uses just 17 dB of global NFB.
Has anyone considered using specialized amps for loudspeaker drive?
By that, I mean using a power amp for say the midrange driver (assuming a 3 way speaker), with classic analog first order filtering located at its input? In fact making it suitable for midrange reproduction only? Series low pass + high pass using simple passive components?
I realize first order filters are nothing to write home about, but in return, they have the least phase shift. You can't have it all.
For the bass and treble amps, you can be a little more ambitious and use the input stage as a second order filter, much like W. Marshall Leach used a second order filter on his first amp way back in the 70ies (in his case, it was used to remove ultrasonic signals).
Just a thought. If feedback can be localized to each stage, why not this?
By that, I mean using a power amp for say the midrange driver (assuming a 3 way speaker), with classic analog first order filtering located at its input? In fact making it suitable for midrange reproduction only? Series low pass + high pass using simple passive components?
I realize first order filters are nothing to write home about, but in return, they have the least phase shift. You can't have it all.
For the bass and treble amps, you can be a little more ambitious and use the input stage as a second order filter, much like W. Marshall Leach used a second order filter on his first amp way back in the 70ies (in his case, it was used to remove ultrasonic signals).
Just a thought. If feedback can be localized to each stage, why not this?
Hi,
I have done such things using Tubes.
In one case the system consisted out of two ESL Panels (appx. 100Hz - 500Hz and 500Hz to above 20KHz) per channel plus a dual 18"Dipole Sub.
The crossover filters and EQ was designed into the Amplifier, which where class A DHT Push-Pull, 300B for the Upper Mid / Treble and 845 for the upper bass / lower mid, non-lopped feedback of course. The Bass Amp was a 1KW Mosfet Module with the lowpass designed in as Sallen Key filter and the dipole EQ in the feedback loop.
The Tube Amplifiers employed normal output transformers and could be switched full-range & flat for driving more conventional speakers, but in system they drove the ESL Stators directly via capacitive coupling from the Anodes.
High pass filters combined the interstage coupling cap's and others to form a 3rd order HPF, while the lowpass & eq was done first order (assymetric slope XO).
If your output stage is Class AB you will need some NFB.
So far my work leads me to believe that the correct number for NFB around a optimum or overbiased Class AB Output stage is around 12dB, this holding for Tubes as much as for Transistors.
More seems to lead to less favourable harmonic distribution and transient behaviour in technical terms and to a more edgy and constipated, subjectively, as NFB ratio's are increased, less seems to cause a sound that is overly soft & wooly and "inconsistent with level". My experimentation remains limited though and other means (like Feed Forward Error Correction) may be employed instead of NFB (and IME with benefit).
Class A Amplifiers can use local degeneration and/or loops only without negative issues, IME.
Ciao T
I have done such things using Tubes.
In one case the system consisted out of two ESL Panels (appx. 100Hz - 500Hz and 500Hz to above 20KHz) per channel plus a dual 18"Dipole Sub.
The crossover filters and EQ was designed into the Amplifier, which where class A DHT Push-Pull, 300B for the Upper Mid / Treble and 845 for the upper bass / lower mid, non-lopped feedback of course. The Bass Amp was a 1KW Mosfet Module with the lowpass designed in as Sallen Key filter and the dipole EQ in the feedback loop.
The Tube Amplifiers employed normal output transformers and could be switched full-range & flat for driving more conventional speakers, but in system they drove the ESL Stators directly via capacitive coupling from the Anodes.
High pass filters combined the interstage coupling cap's and others to form a 3rd order HPF, while the lowpass & eq was done first order (assymetric slope XO).
If your output stage is Class AB you will need some NFB.
So far my work leads me to believe that the correct number for NFB around a optimum or overbiased Class AB Output stage is around 12dB, this holding for Tubes as much as for Transistors.
More seems to lead to less favourable harmonic distribution and transient behaviour in technical terms and to a more edgy and constipated, subjectively, as NFB ratio's are increased, less seems to cause a sound that is overly soft & wooly and "inconsistent with level". My experimentation remains limited though and other means (like Feed Forward Error Correction) may be employed instead of NFB (and IME with benefit).
Class A Amplifiers can use local degeneration and/or loops only without negative issues, IME.
Ciao T
I have a cheap ( Behringer) digital crossover because it is easy for quick testing on the bench for designing speakers. It is no where near a high fidelity unit. Not even close to their $100 analog ones which are surprisingly not as bad as one may think. Not what one can build though.
All drivers are non-ideal. Just some are a lot more "non" than others. In a perfect world, we could do all the filtering on the original bitstream before it reaches a D2A. Someday I will look at Bodzio's Ultimate Crossover. A big PC and several good DAC's does have some attraction. ( I use SoundEasy for speaker design) There is an entire world there to be looked at.
Yes, first amp is the modified Hafler 120, second Rotel 951. LTSpice models attached. Some liberties in devices due to limited models.
The above is purely LTSpice, not reality, and sure not using ones ears. So distortion added by an inductor is what the model does. When I model just a passive filter, I also get quite high distortion. What the model is assuming I have no idea.
It is completely practical for a DIY to design the speaker and amp together. The market has shunned powered speakers probably because so few were done well and if they were, the sticker shock would kill it. So this forces building amps that are immune to the load. Easier said than done. Powered subs seem to be the exception in market, but to my ears, none have been done well.
A lot of designs take some feedback before the outputs in that the drivers are not just tied together but the center point also tied to the output. Something I can plan on plying with when I build my test mule. The models seem sensitive to the driver current and the value of this resistor. I guess I am not following how to add local feedback to the outputs themselves. Larger emitter resistors? ( I am making the leap here that most consider DF to be of little importance) Some have tried to take the feedback after the output network including any fuse that may be there. I guess the extreme of that would be to sense at the speaker terminals. Maybe in a powered speaker, but would sure be looking for trouble in a conventional system. I thin I would only try it with disposable drivers and amps until some very aggressive protection systems were built.
This all comes back to the core of this thread. What makes it SOUND better, and then can we characterize it with a measurement? That is why I posted the differences in shape of the FFT as possible candidate rather than some absolute single number. I have no problem if the measured harmonics go from .0001 to .01 if it sounds better. (or 1% for those fond of glass) I am still leaning toward high order harmonics causing tweeter issues. I have parts on order to test adding steeper LP filters to the tweeters in my current build. (Seas T27-bfc/g) They break up at about 25K as do the tweeters in my Studio-20's. The Vifa's in my last pair break up even lower, and yes, my wife dislikes them more. The high energy is in the 2 to 4K range, so it is not that far a reach for the 5th and 6th harmonic being an issue. I am building two versions of the crossover, cheap parts vs. expensive parts for the tweeter side.
All drivers are non-ideal. Just some are a lot more "non" than others. In a perfect world, we could do all the filtering on the original bitstream before it reaches a D2A. Someday I will look at Bodzio's Ultimate Crossover. A big PC and several good DAC's does have some attraction. ( I use SoundEasy for speaker design) There is an entire world there to be looked at.
Yes, first amp is the modified Hafler 120, second Rotel 951. LTSpice models attached. Some liberties in devices due to limited models.
The above is purely LTSpice, not reality, and sure not using ones ears. So distortion added by an inductor is what the model does. When I model just a passive filter, I also get quite high distortion. What the model is assuming I have no idea.
It is completely practical for a DIY to design the speaker and amp together. The market has shunned powered speakers probably because so few were done well and if they were, the sticker shock would kill it. So this forces building amps that are immune to the load. Easier said than done. Powered subs seem to be the exception in market, but to my ears, none have been done well.
A lot of designs take some feedback before the outputs in that the drivers are not just tied together but the center point also tied to the output. Something I can plan on plying with when I build my test mule. The models seem sensitive to the driver current and the value of this resistor. I guess I am not following how to add local feedback to the outputs themselves. Larger emitter resistors? ( I am making the leap here that most consider DF to be of little importance) Some have tried to take the feedback after the output network including any fuse that may be there. I guess the extreme of that would be to sense at the speaker terminals. Maybe in a powered speaker, but would sure be looking for trouble in a conventional system. I thin I would only try it with disposable drivers and amps until some very aggressive protection systems were built.
This all comes back to the core of this thread. What makes it SOUND better, and then can we characterize it with a measurement? That is why I posted the differences in shape of the FFT as possible candidate rather than some absolute single number. I have no problem if the measured harmonics go from .0001 to .01 if it sounds better. (or 1% for those fond of glass) I am still leaning toward high order harmonics causing tweeter issues. I have parts on order to test adding steeper LP filters to the tweeters in my current build. (Seas T27-bfc/g) They break up at about 25K as do the tweeters in my Studio-20's. The Vifa's in my last pair break up even lower, and yes, my wife dislikes them more. The high energy is in the 2 to 4K range, so it is not that far a reach for the 5th and 6th harmonic being an issue. I am building two versions of the crossover, cheap parts vs. expensive parts for the tweeter side.
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