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Increase (electrical) Damping of a Driver

Are there any commercial amplifiers that have negative output impedance at low frequencies?
Yes. Quite a few active subwoofers use or have used this technique for decades (one pioneering company is Ace-bass, see their 1978 AES paper https://www.statco.it/Docs/AES 1978 Stahl_ACE_Brevetto.pdf), and some active multiway speaker also have amps with non-zero output impedance which maybe partially negative at low frequencies but then returning to zero or more likely larger positive values at higher frequencies because that is what most woofer and midwoofers work best with.
 
The amp will give you this.
An extreme damping factor will immediately remedy this fenomenon. Any out-of-signal movement will simply be cancelled by the amp inverting the back EMF caused by said movement.
Post #6 shows the opposite.

No change in driver decay between damping factor 0.5 and 200.
 
Post #6 shows the opposite.

No change in driver decay between damping factor 0.5 and 200.
Tweeter at 12kHz is no territory for electrical damping. The example was badly chosen.
The driver's terminal voltage is dominated by static VC impedance times drive current, the microphone voltage (aka Back-EMF) is very small in comparison, actually negligible.
For more details: https://www.diyaudio.com/community/...crossovers-in-2025.433915/page-7#post-8152227, especially the third post linked.
 
Tweeter at 12kHz is no territory for electrical damping. The example was badly chosen.
The driver's terminal voltage is dominated by static VC impedance times drive current, the microphone voltage (aka Back-EMF) is very small in comparison, actually negligible.
For more details: https://www.diyaudio.com/community/...crossovers-in-2025.433915/page-7#post-8152227, especially the third post linked.

Okay, then let's use current source amplifiers to drive mid and high frequency transducers while for low frequency transducers, use voltage source amplifiers with negative output impedance at low frequencies in an active loudspeaker system. The transducer model shown below is based on W. Klippel's nonlinear transducer model, https://www.klippel.de/fileadmin/kl...rature/Papers/Klippel_Nonlinearity_Poster.pdf.
screenshot.359.jpg
Note that the AC Reluctance Force, Fr, is related to the AC B magnetic field from the voice coil Induction and Fr also varies with frequency, f. Additionally, due to Sd(x) variation typically related to surround topology (unless you are Purify), Mms does vary slightly with displacement, x; however, Mmd does not. This assumes the diaphragm is a piston with no bending modes.
 
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Tweeter at 12kHz is no territory for electrical damping. The example was badly chosen.
The driver's terminal voltage is dominated by static VC impedance times drive current, the microphone voltage (aka Back-EMF) is very small in comparison, actually negligible.
For more details: https://www.diyaudio.com/community/...crossovers-in-2025.433915/page-7#post-8152227, especially the third post linked.
Yes a maybe not obvious point to add here. If you increase the overall damping factor by modification of the driver then you automatically increase its rise time too - you can see that from a step response of the raw driver/cabinet.

If the goal is better transient response, then independent equalization of phase and amplitude using an FIR filter is your friend.
 
Yes a maybe not obvious point to add here. If you increase the overall damping factor by modification of the driver then you automatically increase its rise time too - you can see that from a step response of the raw driver/cabinet.
If the increased damping has changed the "raw" frequency response then of course the step response changes accordingly. But if you EQ to the same driver terminal voltage vs frequency, step response is the same -- as it must be, despite the different damping.
The effect of the different damping with no other changed variables is in the "fine-print" (like distortion, response to external excitation, excursion overload recovery, etc), as stated in the linked posts of mine.
 
If the goal is better transient response, then independent equalization of phase and amplitude using an FIR filter is your friend.
When used to remove excess phase like from any classic XO, then yes.
But if you bend phase away from already minimum phase, the results are mixed. Typical example would be the full compensation of the system highpass function, say, 6th order (4th order from a ported alignment + 2nd order a from subsonic filter), this doesn't work in practice, too much pre-signal. Partial phase roll-back to, say, 3rd or 4th order order can work, with tolerable pre-signal artifacts.
 
When used to remove excess phase like from any classic XO, then yes.
But if you bend phase away from already minimum phase, the results are mixed. Typical example would be the full compensation of the system highpass function, say, 6th order (4th order from a ported alignment + 2nd order a from subsonic filter), this doesn't work in practice, too much pre-signal. Partial phase roll-back to, say, 3rd or 4th order order can work, with tolerable pre-signal artifacts.
Do you know of any studies on the audibility of pre-ringing? I know from studio work that is audible, that's without question, but I don't know of any decent guidelines.
 
Do you know of any studies on the audibility of pre-ringing? I know from studio work that is audible, that's without question, but I don't know of any decent guidelines.
This study is pretty convincing to me


I recently equalized my KEF R900 woofer system using a FIR filter and can honestly say the audiable results are very pleasing to my ear. The transient performance is much improved. I dont see any pre-ringing of the acoustic impulse response at all - thats with a 48dB/OCT acoustic roll-off. The filter is of course not linear phase.
 
Do you know of any studies on the audibility of pre-ringing? I know from studio work that is audible, that's without question, but I don't know of any decent guidelines.
I'm not aware of studies that look at linear-phase vs minimum-phase high-pass functions at very low frequencies (like well below 100Hz) but I did my own little private research.
@Waveform Fidelity , thanks for the link.

The pre-signal effect of a symmetric linear-phase step response like this (6th-order Linkwitz-Riley high-pass at 30Hz) is very audible when listening to the step, and with real music signal like kick drums it is very audible as well. The signal starts like at least 30ms before the main event and has little chances of being masked.
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I don't call it pre-ringing because it does not sound like ringing at all, rather it sounds almost like an increasing noise burst with a brown'ish spectrum. Kick drums or plucked upright bass notes sound a bit like if they were preceeded with mirrored copy in time (well, they are actually, sort of), making them quite artificial. While group delay is now flat and that aspect is clearly audible as in that the low bass is not delayed anymore, that quirky time-stretching quality totally spoils it.
 
I'm not aware of studies that look at linear-phase vs minimum-phase high-pass functions at very low frequencies (like well below 100Hz) but I did my own little private research.
@Waveform Fidelity , thanks for the link.

The pre-signal effect of a symmetric linear-phase step response like this (6th-order Linkwitz-Riley high-pass at 30Hz) is very audible when listening to the step, and with real music signal like kick drums it is very audible as well. The signal starts like at least 30ms before the main event and has little chances of being masked.
View attachment 491150
I don't call it pre-ringing because it does not sound like ringing at all, rather it sounds almost like an increasing noise burst with a brown'ish spectrum. Kick drums or plucked upright bass notes sound a bit like if they were preceeded with mirrored copy in time (well, they are actually, sort of), making them quite artificial. While group delay is now flat and that aspect is clearly audible as in that the low bass is not delayed anymore, that quirky time-stretching quality totally spoils it.
I think it's called preringing solely because of the time domain measurements used to diagnose the issue. The sound in higher frequency transients with linear phase EQ (I've never heard it with crossovers specifically) is like a noisy, flanging, reversed version of the original.

Do you have any test signals you can share?
 
I'm not aware of studies that look at linear-phase vs minimum-phase high-pass functions at very low frequencies (like well below 100Hz) but I did my own little private research.
@Waveform Fidelity , thanks for the link.

The pre-signal effect of a symmetric linear-phase step response like this (6th-order Linkwitz-Riley high-pass at 30Hz) is very audible when listening to the step, and with real music signal like kick drums it is very audible as well. The signal starts like at least 30ms before the main event and has little chances of being masked.
View attachment 491150
I don't call it pre-ringing because it does not sound like ringing at all, rather it sounds almost like an increasing noise burst with a brown'ish spectrum. Kick drums or plucked upright bass notes sound a bit like if they were preceeded with mirrored copy in time (well, they are actually, sort of), making them quite artificial. While group delay is now flat and that aspect is clearly audible as in that the low bass is not delayed anymore, that quirky time-stretching quality totally spoils it.
Is the waveform of the woofer SPL or the electrical signal at the FIR filter output?
 
I think it's called preringing solely because of the time domain measurements used to diagnose the issue. The sound in higher frequency transients with linear phase EQ (I've never heard it with crossovers specifically) is like a noisy, flanging, reversed version of the original.

Do you have any test signals you can share?
Its because there is some signal occurring before the main impulse - the amplitude of which depends upon the roll-off rate of the filter. Brick wall filters are an extreme case which are used often as an example of pre-ringing from the ideal filter, but the acoustic pre-ringing, the SPL, is nothing like this in a practical case. To flatten the amplitude and keep the phase constant within a woofer intended pass band would take many hundreds (possibly a thousand or so) of FIR filter taps and that results in too much delay in the filter (100’s of milli-seconds), so designers compromise and accept less than flat amplitude and less that constant group delay at the expense of filter delay, but reap the benefits “bass punch” improvements from improving the group delay. Engineering is always about compromise. From what I have seen on audio forums, the topic is often misunderstood.

If you want to learn about pre-ringing, I invite you to research “Gibbs Phenomenon” - diagnosis is not needed as this is standard signal processing theory.

In reality there are very few use-cases where brick wall filters are actually needed in crossovers.
 
Is the waveform of the woofer SPL or the electrical signal at the FIR filter output?
It shows the acoustical target. The FIR output is that of a time-reversed allpass with the same phase response as the 6th-order highpass so the combination has net zero phase.

Do you have any test signals you can share?
A synthetic signal to test the effect of linear-phase bass roll-off, you mean?

n reality there are very few use-cases where brick wall filters are actually needed in crossovers.
I tend to agree. The Horbach-Keele type is maybe the most prominent case that uses infinitely steep final roll-off. But the transition region still is quite wide and rounded, reducing ringing.

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We've gotten waaay off-topic by now, we should move on or discuss these matters in a separate thread, I'd think.
 
Its because there is some signal occurring before the main impulse - the amplitude of which depends upon the roll-off rate of the filter. Brick wall filters are an extreme case which are used often as an example of pre-ringing from the ideal filter, but the acoustic pre-ringing, the SPL, is nothing like this in a practical case. To flatten the amplitude and keep the phase constant within a woofer intended pass band would take many hundreds (possibly a thousand or so) of FIR filter taps and that results in too much delay in the filter (100’s of milli-seconds), so designers compromise and accept less than flat amplitude and less that constant group delay at the expense of filter delay, but reap the benefits “bass punch” improvements from improving the group delay. Engineering is always about compromise. From what I have seen on audio forums, the topic is often misunderstood.

If you want to learn about pre-ringing, I invite you to research “Gibbs Phenomenon” - diagnosis is not needed as this is standard signal processing theory.

In reality there are very few use-cases where brick wall filters are actually needed in crossovers.
My comment was specifically responding to @KSTR saying that preringing doesn't sound like ringing despite the name.
A synthetic signal to test the effect of linear-phase bass roll-off, you mean?
Yes, exactly.
 
The problem is that the DC resistance of the voice coil was totally ignored. For "electrical damping", the effects from the amplifier output resistance and the voice coil DC resistance are the same. Once the amplifier output resistance is below about 10% or 20% of the voice coil resistance, reducing it further does nothing (in terms of electrical damping). This has been pointed out by George Augspurger almost 60 years ago (source), 8 years before Dr Toole's article "Damping, Damping Factor, and Damn Nonsense" (source).

View attachment 489290
There seems no denying that, by two of the giants who's shoulders we still stand on today, no less.

Could something else (also) be at play, since Putzeys' class D's have practically load invariant (flat) frequency response as well? And they're known for Bruno's ample use of negative feedback loop "taken only at the speaker output". All this in the realm of active amplification of course, as opposed to having load-, temperature- and frequency-variant XO components in the loop.

Why do his units rule, and are found inside so many reputable (up to the crazy expensive) high end amps?

I'm no circuit genius obviously, but AFAIK all basically well designed amps, as giant Bob Carver proved (should) sound the same: they have no sound of their own. As all other audio components except loudspeakers and rooms don't. Yet when i used some of Putzeys's UcD's i heard something clearer and more articulated than ever before.

An admittedly very subjective benchmark, but it was audible, and not just audibly different but audibly better! Would have to measure to know what made it so, but better for sure. What could be going on, apart from me being a dumb ass?
 
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Would have to measure to know what made it so, but better for sure. What could be going on, apart from me being a dumb ass?
Most likely perceptive bias. You simply cannot separate this from uncontrolled sighted listening.

And if you measured you would see measurable differences - but you would have no way of knowing if these were enough to cause an audible difference unless you did properly controlled and blind listening comparisons to prove you are hearing a genuine audible differences in the sound waves reaching your ears - rather than the stuff that is made up in the wetware between them.
 
Even though i'm a little dumb, i AM aware of all that. And you're essentially right of course.

Still, this was too explicitly noticeable to be attributed to (only) that.

Then again, Toole did mention that the DF was more of a side effect of the distortion containment through feedback good amps feature, and Bruno uses that liberally too.

Anyway.
 
Still, this was too explicitly noticeable to be attributed to (only) that.
Why do you think perceptive bias can't create explicitly noticeable differences? It is very capable of doing so - unlike the levels of distortion created by modern well measuring amps. In fact I would revers your statement. "This was too explicitly noticable to be distortion - more likely to be bias"

Though i did forget to include the other very likely possibility for your perceived differences - and that is insufficiently accurate level matching, using a volt meter at the speakers with a test tone - and matched to within 1 to 2 % (aprox 0.1 to 0.2 dB)
 
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