I didn't cut your post... it's too interesting....
You brought out two very interesting aspects of audio/acoustic reality:
The harmonic structure of distortion... Let me call this HSD for short.
Output level affects the HSD in a
Thus, to minimize the change of HSD, we want to operate the output devices in the narrowest practical range of its operating range.
This is why we like to listen to class A amplifiers that are biased at such crazy levels.... when the output device sits at 30V and is required to swing only 20V... from 40 to 20 V... with sufficient hear room above and below... it is acting in its most linear range. Hopefully in its linear range, it preserves the harmonic structure of its distortion.
Also, if the output device is asked to put out not too much power, meaning say it sits biased at 20V put it only puts out +/- 3V.... say... that means that the device is stying withing a narrow range of its operating environment.... hence, the most consistent HSD.
You know... maybe Nelson Pass is actually right when he says The First Watt is what counts.
I hate to sound like Steve Martin ( excuuuuuse me.. )...
But perhaps our choice of inefficient speakers is wrong. Perhaps we want 98db efficient speakers that only require an amplifier biased so it dissipates 300 watts but only drive 10 wpc into the speaker?
Since Harmonic distortion is measured, roughly, as THD, maybe that's the trick. Don't bother (within reason) with the level of THD, just make sure the harmonic structure of the distortion is constant through the operating range.
Not that simple. If not using synchronous sampling and not discarding phase information then averaging multiple acquisitions doesn't work as you might expect. Seems to me cancelation of the test signal is possible since its correlation with, say, cos 2f may be +1 on one acquisition and -1 on the next acquisition depending on the exact phase of the test signal at the instant the acquisition happens to start. That averages to zero.
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Simplify, a 3 way loudspeakers with 2nd order paper drivers, high sensitivity, 10 inches woofer ported, that has inherent hight distortion, then you install with the same amp a smaller harbeth with lot less sensitivity and it plays with less distortion by a good margin.
The power of the amplifier is of zero importance, as well as it's distortion amount.
Let me go in deep, the fact that another amp has a 6db margin of more power (200 watts amp vs 50), has no value since the 50 watts amp sufficiently drive way beyond the drivers capacity of excursion, the subjective listening gain of quality is relatively small.
The quality of the amplification is paramount to power.
I would suggest the fact that the 50w amp has enough power to send the drivers to their full excursion - and so 200w is simply not needed - is not the full story.
You forget about transients - which people typically say can demand 10x as much power (as average listening requires) to reproduce perfectly. So if you are using 5w at normal listening ... a transient might demand all of the 50w amp's power - whereas the 200w amp still has plenty of headroom.
This is where amplifier conception is lacking. For transients, delivering mostly peak amps which drivers are eating like sumo.
First of all the power of a badly designed 500 watts amp can turn pretty smallish into capacitive loads, what can it really deliver through the xo to the voice coil, can it drive reactance, capacitance loads? It is not ingeeneered for that.
The other elephan is that class A wasting nearly all the current of the power supply into a pancake cooker , it's already turning at 6000 rmp so when you want that transient it is 3/4 exhausted, on the other hand , the low class A can turn the juices very fast and add the extra 10 amps readily available and quickly return to its non stressed operating point in no time.
Tube's can't do that so the output transformers need to be of relatively big size with hundreds of Henry's inductance, they reach half of transistors capacity roughly.
For clarification, the power/thd rating says nothing about the ability to send quick surge of a dozen amps into a capacitance/reactance load, a 20 watts ss amp could be designed to temporary send 30 amps , but for continuous signals no more than 1 amp, , my main amp can draw 6 amps of mains for short periods, it it still a low power amp.
First of all the power of a badly designed 500 watts amp can turn pretty smallish into capacitive loads, what can it really deliver through the xo to the voice coil, can it drive reactance, capacitance loads? It is not ingeeneered for that.
The other elephan is that class A wasting nearly all the current of the power supply into a pancake cooker , it's already turning at 6000 rmp so when you want that transient it is 3/4 exhausted, on the other hand , the low class A can turn the juices very fast and add the extra 10 amps readily available and quickly return to its non stressed operating point in no time.
Tube's can't do that so the output transformers need to be of relatively big size with hundreds of Henry's inductance, they reach half of transistors capacity roughly.
For clarification, the power/thd rating says nothing about the ability to send quick surge of a dozen amps into a capacitance/reactance load, a 20 watts ss amp could be designed to temporary send 30 amps , but for continuous signals no more than 1 amp, , my main amp can draw 6 amps of mains for short periods, it it still a low power amp.
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I was thinking about the same thing while smoking my evening cigar....
Yes, for soundstaging we have to take a holistic system approach. And so we have to take into account the speakers too.
We need to ensure that the speakers operate in a linear fashion as well.
Look at Mofi Powersource 10. A coax design. According to its designer, the size of the woofer was chosen because the woofer is a waveguide for the tweeter, so he wanted minimal cone excursion. This means that the woofer is generating acoustic power while operating with minimum excursion... by pure happenstance that also means the cone is operating in its most linear range.
Same thing with the big coax Tannoys... which combine high efficiency with limited cone excursion.
Is it possible then that by pure happenstance, those designers came upon the ideal design for dynamic range, low distortion and a steady soundstage?
The ENTIRE system must operate on its most linear range of operation... to minimize compression artifacts.
Recall how @Zen Mod went on and on a while back about minimizing system gain to the minimum required? That might be part of the solution. Our systems have too much gain.
I've been thinking about intermodulation distortion... more on that... I might need another cigar for that.
Headroom is indeed extremely important.
In order for the SYSTEM to operate in its most linear range, it must have excess headroom. That means that both the speakers and the electronics must be operated in a narrow range of very linear operation with gobs of headroom.
Perhaps we need 14 inch woofers that move back and forth 1/16 of an inch? With a horn loaded coax tweeter?
But we still need to pay attention to how the harmonic structure of distortion and the intermodulation comes into play. This means a system with very low levels of harmonic distortion because IM is caused when harmonics interact.
So, a system with very low harmonic distortion is required through the entire range of operation for both the electronics and the speakers. So devices with sufficient power operated at very low output into very high efficiency speakers that barely move.
Or extremely powerful amplifiers, 1000 watts, operated at a fraction of what they can put (for the lowest distortion) into planar speakers?
And then you got the noise level... hmm....
It's a saturday... I got time for another cigar. See ya tomorrow.
BTW, let me note, as I grab my cigar, that the Maggie 1.7 driven by the Iron Pre Bal->BA3B->bridged F4s sound beautiful at all levels. The very low noise allows dynamic operation at low levels... meaning the entire system is operating way into its least distortion mode. Imagine... 200 watts of amp power operating at The First Few Watts Mode (*) into a pair of Maggies. And it sounds.... incredibly realistic!
(*) when you run Maggies, it's the First Ten Watts that matter... 🙂
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I know Maggies extremely well, Tony ... I had big 3-ways (with the 'true ribbon') for over 25 years. 🙂
Running them active allows them to sing - without needing to use a Sanders Magtech (500w into 8 / 900w into 4!). For most of the time I had them, I used:
* 180w into 4 ohms on the bass panels
* 120w into 4 ohms into the 3 ohm ribbons
* and the same for the 2 ohm ribbons.
All Hugh Dean's 'AKSA' amps.
Yes, the larger Maggies sound excellent... but they need a bigger room than my 14x20x8....
You should hear the smaller 1.7 with a 10wpc Sissy SIT... They have no right to sound soo good. not loud, mind you... but since my system's noise floor is so low, you still hear dynamics.
Sure, the bridged F4 amps have a bass slam that you have to hear to believe... I think the bridged F4s have a bit "extra" bass in there... but it really matches the speaker.
Why did you get rid of them? What are you driving now? This? https://www.aesmelbourne.org.au/wp-content/media/THE AKSA STORY.pdf
But we digress from the THD discussion... well, except that Maggies are low distortion speakers.
You should hear the smaller 1.7 with a 10wpc Sissy SIT... They have no right to sound soo good. not loud, mind you... but since my system's noise floor is so low, you still hear dynamics.
Sure, the bridged F4 amps have a bass slam that you have to hear to believe... I think the bridged F4s have a bit "extra" bass in there... but it really matches the speaker.
Why did you get rid of them? What are you driving now? This? https://www.aesmelbourne.org.au/wp-content/media/THE AKSA STORY.pdf
But we digress from the THD discussion... well, except that Maggies are low distortion speakers.
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Indeed - they sound better with space!
My last room was 26' x 17' - with a 'cathedral' ceiling which was about 15' high at the central ridge line.
The Maggies sounded really great in there - but 10 years ago, we downsized into a new house where the 'music room' was about your size - 12' deep x 19' wide x 9' high. (Spkrs on a long wall.)
In this room, the Maggies were constricted - and they dominated the space! So, after a few years, I promised my wife I would replace them with some spkrs which had much less of a visual impact on the room. But of course ... they had to sound good!
So my new spkrs are 'zero baffle':
- 4x SB Acoustics 6 1/2" Textreme mid/bass drivers
- and the SB Acoustics ribbon tweeter.
I posted a pic earlier but here it is again:
I have to say ... I don't miss the Maggies at all; in fact, driven by the Class A 'winter' amps - these actually sound better, in that room!
Well done for finding that article! 👍 Yes, I've used Hugh's amps all the way through.
https://www.aesmelbourne.org.au/wp-content/media/THE AKSA STORY.pdf
And their drivers are basically totally resistive - albeit around 4 ohms. This makes them a relatively 'easy' load - except they are so ridiculously inefficient. 🙁
These are diy 7 inch, 2 ways, They have less THD and NOISE than reference commercial loudspeakers (5k), they are almost finished. In the test 100hz thd bump is the room, they are close to wall and floor, but at 1m it is acceptable, I used a reference calibration microphone. In Green is a 6bl7 \\ push pull amp with quite a soft power supply with high impedance, in brown is the SS amp which has at least 3 times the wattage , similar THD profile very low output Z, and 10 times at least the current drive. I am not a specialist but I can't explain the phase which is way better on the tube amp, the THD proves that the power doesn't matter, I drove the woofer at almost double its max excursion, the impulse seems way better in the SS amp which has a very high slew rate, low feedback, the extension is voluntarily cut at 20khz , but it extends easily in the 30+ (note -50db is 0.3%thd roughly, first graph is THD in db)
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Listening to them both, I can say the THD or power of both amp is plenty adequate, and what I hear as difference (the Fr. response is the same), is the phase and driving impulse as well as noise floor during woofer control. The extra 1% thd reduction at 120db of a 1000watt amp would not matter at all at this point the woofer would almost cook and produce way more THD, actually a 10 Watts amp is preferable because it will not destroy my drivers and gently clip protecting the loudspeakers. (sens is 85.6db 1m 2.83v)
How about testing for horizontal localization? For ITD below 1.5kHz a VST sample delay plugin can be used to position a mono instrument sound recorded on two stereo channels. Delay one channel enough to incrementally move the instrument location between the speakers from center over to one side or the other. Here is a free VST sample delay plugin that can be used: https://socalabs.com/effects/sampledelay/ It can work with any DAW.
Then use volume panning in the DAW to do similar tests with frequencies above 1.5kHz for localization testing by ILD.
Beside horizonal location, try to hear how wide in the horizontal axis is the instrument or test tone image perceived to be. Does it sound, say, 1" wide (much like a point source), 12" wide, 3' wide, or something else? To put it another way, how many virtual sound sources could you put from left to right across the sound stage while keeping them sounding fully separated from each other? Could you distinctly hear each drum in a drumset being at a different horizonal location?
If the answer to the above sorts of tests is that that imaging is rather vague, maybe time to look at your dac.
Then use volume panning in the DAW to do similar tests with frequencies above 1.5kHz for localization testing by ILD.
Beside horizonal location, try to hear how wide in the horizontal axis is the instrument or test tone image perceived to be. Does it sound, say, 1" wide (much like a point source), 12" wide, 3' wide, or something else? To put it another way, how many virtual sound sources could you put from left to right across the sound stage while keeping them sounding fully separated from each other? Could you distinctly hear each drum in a drumset being at a different horizonal location?
If the answer to the above sorts of tests is that that imaging is rather vague, maybe time to look at your dac.
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It seems the phase delay of the tube amp replaces the phase at the proper place...., such a weird thing to see,
This simply looks like the green amp is inverted and delayed. Try the measurement with reversed polarity. No idea what's the delay component - different mic position, not the same measurement session?
Had a 2nd look at the phase - it is 180° different (e.g. 200Hz). So it's pretty sure one of the amps is wrong in polarity!
But there still seems to be a delay ... can you repeat these measurements, at least with one amp with switched polarity so we can have a closer look about the delay issue?