You are VERY VERY wrong, and ignoring the tens of MILLIONS of analog/transistor/OpAmp/FET/you-name-it SS Guitar Amplifiers which have been designed and made since the 60's.
Personally I have been designing and making SS Guitar amps since 1969, go figure, over 14.000 of them.
Have yet to fit a single digital gate inside any of them, let alone their overblown offspring, the Microprocessor.
And all, including the Digital ones, are trying to emulate the Tubes, which are as analog as can be, by definition.
I get a couple laughs a Month when kids set their 15W 8" microprocessed/DSP equipped amps to the "Jimi Hendrix at Woodstock" or "Metallica at Monsters of Rock" setting and expect even 1% of the sound experience the real one has.
Hint: it takes at least 4 to 12 EL34 equipped amps, driven 20dB or more beyond clipping, driving 32 to 96 12" Celestion speakers for that.
Hint 2: set one of those "16 sound" or "64 sound" wonders loud (I mean as loud as they can which of course is "nothing" 😉 ) and go through all the available settings.
All you´ll get is 16 or 64 slight variations of telephone on the table Fuzz.
Practical experience: every time I´m at some multi-band Festival, invariably one or more Guitar Player plugs his breakfast table sized pedalboard (Digital of course) into a standard SS combo amp (60-120W ones driving a single or dual 12" are very popular) , smiling and confidently claiming "MY sound is in here" and then proceeding to put out nasty buzzy muddy excuse of a sound.
I´m happy that at least a few notice something is very wrong and think the pedalboard or amp are broken .... "it didn´t sound this way at home".
Welcome to the stage sound level World, Baby, no digital trickery will save you, if you don´t have the muscle to push air, lots of it.
At that particular point in the discussion it was being steered toward cramming the output of the pickup straight into the ADC. In that use case, all of the “sound” would be accomplished with “digital trickery” and you could just buy the latest Tripath chip for the power stage. Which you might be able to turn down low enough to not get evicted. Or you could buy an iNuke amp and be heard from space. You couldn’t do that in the 60’s. And a really good solid state or tube amp from the 60’s turns out to be a darn inconvenience being crammed into an apartment these days. I like my amps loud too but there is not always a place for that.
I'm sorry, but no, that is not even the slightest bit correct. Not even the teeniest, tiniest, weeniest, little bit!
Consider:
Powerful magnet + low Q does not lower THD caused by nonlinear speaker surround.
Powerful magnet + low Q does not lower THD caused by nonlinear voice coil spider.
Powerful magnet + low Q does not make frequency response almost independent of sealed speaker cabinet volume.
Powerful magnet + low Q does not make speaker frequency response virtually independent of room temperature.
Powerful magnet + low Q does not automatically compensate for manufacturing variations in loudspeaker sensitivity.
Powerful magnet + low Q still does not make the voice coil motion accurately track the incoming electrical signal.
To understand the difference at gut level, try this simple experiment:
Find a notebook with lined pages. Turn on an audio book of your choice, let's say The Adventures of Huckleberry FInn. Close your eyes, and write down one page of what you hear, trying to stay between the lines, with your eyes closed the whole time. This is what happens when you have no negative feedback.
Now open your eyes, go back to the same paragraph in your audio book, look at your writing as you write, and once again write down one page of what you hear, trying to stay between the lines.
How well did you stay between the lines in the first instance (eyes closed)?
How well did you stay between the lines in the second case (eyes open, looking at what you're writing)?
The second time, you had negative feedback from your eyes to your brain, telling you every time your hand drifted even a little bit away from where it was supposed to be. The negative feedback allowed you to re-position your hand, correcting for errors as you went.
The first time, with eyes closed, you had no negative feedback. You were relying on the open-loop behaviour of your hand/brain control system.
If you are like every other human being whom I've encouraged to try this test, you will find that your hand does not stay between the lines very well when you don't have negative feedback from your eyes.
Without motional feedback, a loudspeaker cone is "blind" to the audio signal, just as you were blind to how well your writing stayed between the lines.
With motional feedback, the loudspeaker cone "knows" what it's supposed to do, and corrects errors as it goes, just like you did when your eyes were open.
Yes, as different as driving down a twisty country road with a blindfold, vs driving down the same road with your eyes open on a clear sunshiny day!
And that is a decent analogy, not just hyperbole one my part. 🙂
Most of the worst sounding speakers I've ever heard have been compression drivers. Both the driver itself, and even more the horn, are subject to tremendous errors in frequency response.
Compression drivers with attached horns are more efficient, yes. Therefore they are still used when loudness matters more than good quality sound. If you want very loud, but usually very crappy sound, over a limited bandwidth, a compression driver connected to a horn is a good way to get it.
No, the trouble with horns is much more basic than that. That they work wonderfully in simplified mathematical models, where the loudspeaker has zero diameter, and the horn has infinite length, and infinite mouth size, and the exponential flare is absolutely perfect, and the air is absolutely linear even under high SPL.
But real-world horns and tweeters almost invariably result in horrid frequency response, and poor quality sound. You can blame the compression tweeters for the nasty sound quality from most affordable PA speaker systems.
I Googled for a random compression tweeter frequency response, and a random soft dome tweeter frequency response. The first images I came up with are attached below. The horn is a Fostex 025H27. The dome is a CSS LD22F. I have no previous familiarity with either model.
The horn has an incredibly poor frequency response, fluctuating irregularly over a range of at least 7 dB (!) over its narrow usable frequency range of between 1.5 kHz and 12 kHz.
Over the same frequency range, the dome tweeter has an almost ruler-flat frequency response, maybe varying by +/- 0.5 dB at the worst.
Not only does the horn have a nasty frequency response and limited frequency range, it doesn't even live up to expectations of efficiency. The dome tweeter is actually about 5 dB louder at the same input power over the entire frequency range! The horn manages to equal the dome tweeter only at the worst peak in its very irregular frequency response, at about 4.5 kHz.
I guarantee that this horn will sound nasty no matter what you do with it. I guarantee that this soft-dome tweeter will sound good if used properly.
I'm deeply biased against compression drivers, for good reason. Most of them are just as awful as this crappy Fostex one. Listening to compression tweeters with very irregular frequency responses like this one, especially at higher SPL levels often found in live performances, often causes me physical pain, like being stabbed in the ears with needles.
I have heard compression tweeters in P.A. systems that sounded good, but only when you go way, way up in price, and buy premium quality brands.
-Gnobuddy
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I used to play bass and peaked out with (the now-common) 1kW of amps. It still clipped, at sound levels that were just keeping up with a nominal 12W guitar amp and standard drum kit.
Now, compared to popular home audio speakers, mine were efficient. Just not particularly so for guitar/PA duty (Hoffman's iron law vs the size of my car boot and the size of my wallet).
Secondly, it was all high feedback "perfectly engineered" SS stuff. You know the rest.
Coming back to home audio, I agree with your analysis of most music- it's "mastered" to death. Heck, these days it's got to sound good from a battery powered, beer-can-sized widget after being transferred over a low-res bluetooth link. HiFi it's not.
Gnobuddy -- what's missing in all of what you're saying, is a sanity check. Did you compare the in-room response with 2 speakers -- one with MFB and one without, using a microphone? I'm saying the MFB would cause in-room resonances to be a lot worse.
And I already offered an explanation for how it happens: reflections get boosted precisely because the NFB is flipped and turned into positive feedback. Instead of the in-room sound getting damped by a softly-suspended mass on an air-spring, the driver applies an opposing force, a situation that reduces stability.
Which custom guitar would have the longest sustain?:
Option a,
The strings are tied at one end to a solid concrete wall, and tied at the other end to a solid woofer cone with MFB?
or option b,
The strings are tied at one end to a solid concrete wall, but at the other end a softly suspended pliable mass on an air-spring is used?
And I already offered an explanation for how it happens: reflections get boosted precisely because the NFB is flipped and turned into positive feedback. Instead of the in-room sound getting damped by a softly-suspended mass on an air-spring, the driver applies an opposing force, a situation that reduces stability.
Which custom guitar would have the longest sustain?:
Option a,
The strings are tied at one end to a solid concrete wall, and tied at the other end to a solid woofer cone with MFB?
or option b,
The strings are tied at one end to a solid concrete wall, but at the other end a softly suspended pliable mass on an air-spring is used?
Come on, let's get serious. The domed tweeter response is filtered smoth. The horn should ber crossed over at 3 kHz. these are the old Radioshake horns that I bought back in the day for something like under $10. So many words for a budget speaker. Must be a slow day. Wonder how long the dome tweeter would last with one inadvertent case of feedback in a live situation?
Well, now you talking Bass here.
Beyond sheer SPL, Hi Fi (flat and low distortion) equipment are fine for Bass.
And some amps and matching cabinets (SWR or Trace Elliot) are very Hi fi in sound.
Now Guitar is different: "a Guitar amp is a processor and its speaker a last stage equalizer"
Agreed, to a point. Which is if I use common HiFi design principles, I need a bucketload of headroom as it sounds like poop when it clips. If I use a "clean" (in the Gnobuddy sense of the word) amplifier which clips/limits gracefully I can have 10db less peak SPL capacity and yet (mostly) have the same sonic impact.
Some styles do demand all that head room (basically, the closer we get to just amplifying the acoustic sound of the instrument)
I've been serious the entire time.
Not at all. Why do you assume I would make such a basic mistake?
Both the horn and the tweeter have some little smoothing applied to them, which is standard in all these sorts of loudspeaker measurements, but one isn't smoothed more than the other.
I've measured good dome tweeters. A good one gives you a very smooth response; +/- 1 dB from 1 kHz to 15 kHz is not unusual. The titanium dome tweeter in the original Mackie HR-824 was almost perfectly flat all the way out to 40 kHz, at which point there was a tall, narrow "spike" in the frequency response - we called that the "oil can resonance", where the acceleration forces finally overcame the stiff stiff metal dome and caused it to evert.
As for the horn, in my experience, they are, almost always, just that crappy.
Then you get crap from 3 kHz to 10 kHz, instead of crap from 1.5 kHz to 10 kHz. But it's still crap.
A co-worker used to say "You can't polish a turd". That applies to every affordable horn tweeter I've ever encountered.
ADHD friendly version: Horns measure like crap. They sound like crap. Only use them when loud crap is what you want.
Yes, they are fragile. The question is: Do you want loud, but $hitty, sound quality? Or less loudness in exchange for better sound quality?
-Gnobuddy
Remember, I was working on a floor full of smart audio engineers, who made a living from pro-audio, and had a great deal of collective expertise.
As a matter of routine, every time the loudspeaker group had a prototype worth listening to, we would let everyone on the engineering floor know, either by email or word of mouth, and invite anyone interested to come down and listen, and give us their feedback. There were also some people on other floors (marketing, sales) who were interested, and they would get invited too.
Depending on how busy people were, you might get half a dozen people dropping by, or ten, or twenty.
So of course we didn't miss anything as obvious as basic sanity checks. If one person missed something, there were always at least half a dozen other smart people who would notice, bring it up, and make sure it was implemented.
That was one of the great joys of working with a team of smart, knowledgeable people.
Of course.
I also set it up so that I could switch the MFB in or out, on the same speaker, so it could be auditioned either way. Switching out the MFB changes the frequency response (flattens and extends it), and tightens the bass response (much better transient response). By switching the MFB in or out, we could listen for audible benefits, or not.
It didn't. There was zero evidence that MFB had any effect whatsoever on room resonances. And I don't believe there is any reason to expect the effect you're hypothesizing.
What you actually hear when you switch the MFB in, is much "tighter" bass. Exactly what you expect. The MFB makes the woofer cone follow the incoming bass signal almost perfectly, rather than flopping around in a much less controlled manner, like any conventional woofer.
Didn't happen. Nor did I expect it to. I don't follow your reasoning as to why you think NFB becomes positive feedback, or incites room modes. It doesn't.
Let's talk about that a bit more later in this post.
Nope. Write down the differential equation, solve it, and you'll find that applying an in-phase, opposing force does not cause instability. Not at all.
In fact, we can get a pretty good idea of this without even doing any calculus. Consider f=ma, Newton's most famous equation.
For our purposes, "f" is the force that vibration of a room mode applies to the loudspeaker cone.
Now let's add MFB, which you postulate is going to add an opposing force to the speaker. Mathematically, that means the MFB is adding a force, (-k*f), to the original force, f.
The minus sign takes care of the fact that the feedback opposes the original force. "k" is a constant, a numerical scaling factor, which is set by the amount of MFB being used. k=0 means no MFB. k=1 would be an infinite amount of motional feedback (which is impossible in practice).
The net force on the system is now the sum of (f) and (-k*f), i.e. the net force is (f-k*f).
Pull out the common factor "f", and you can rewrite the net force as f(1-k).
Newton's equation of motion - with NFB applied - is therefore:
f(1-k) = ma
(Instead of the original f=ma).
Remember that k is just a constant, varying between zero (no NFB) and 1 (infinite NFB).
That means (1-k) is also just a constant, varying between 1 (no NFB) and 0 (infinite MFB).
f(1-k) is therefore simply a scaled-down copy of the original force. Whatever "f" does, f(1-k) does, only at smaller amplitude. Like scaling down a sine-wave with a volume pot.
And that, in turn, means that "a" (the acceleration of the mass) is just a scaled-down copy of the original motion. There is no new instability introduced whatsoever.
In fact, this business of applying an opposing force is the basis of every servo-feedback scheme, and they are everywhere, including in your body. This is why your hand doesn't burst into oscillations every time you look at it while you're writing (NFB via your eyes).
The same applies to the electrical output of any amplifier with NFB. Instead of room modes driving it, thermal noise is driving it, trying to "shake" it. Because the amplifier has NFB, the NFB opposes the "shaking". If your argument was correct, amplifiers with NFB would produce much more noise at the output, as the NFB would worsen it, in the same way you expect it to amplify room modes.
But this doesn't happen. Negative feedback in amplifiers was invented by Harold Black in 1927. We've been using it for 95 years, in millions and millions of amplifiers, and they don't become noisier when NFB is applied. Instead, the noise is reduced, in proportion to the amount of NFB. (Apply 40 dB of NFB, and both output voltage and output noise drop by almost exactly 40 dB, leaving signal-to-noise ratio unchanged.)
Getting back to the MFB speaker in a room full of air, as I mentioned earlier, typical speaker materials have a density a thousand times greater than air. Coupling from room air to speaker is negligible. Imagine throwing lots of those little cotton balls women use to apply makeup at your car; the low density of the cotton means the cotton balls don't shake the car (heavy steel) noticeably.
Another thought experiment: Put two identical speakers into a room, let's say at diagonally opposite corners, one on the floor, one just under the ceiling, to create symmetry in all 3 dimensions. Drive 1 watt of audio power into one speaker. Now measure the signal from the other speaker, acting as a microphone.
This gives us an idea how much of the acoustic power in the room actually couples back to electrical output from the second speaker. With 1 watt driving the first transducer, how many watts come back out of the second one?
I haven't done the experiment, but I'm sure the answer would be "A very tiny fraction of the original 1 watt". I doubt you would get even as much as a milliwatt.
So there really isn't much connection between a loudspeaker, and the "Q" of the room air modes. They're extremely weakly coupled to each other. Much like an elephant stomping the ground at the zoo ten miles from your home doesn't shake you on your living-room couch.
-Gnobuddy
Agree, and just want to add that I think the word "clip" often confuses things, because it implies some sort of abrupt amplitude limiting. Vintage valve audio circuits tended to very gently and progressively "squash" the waveform as you turn it up, rather than actually "clip" it. Those in search of clean guitar tone might be a long way away from any actual clipping, but still have a few percent low-order THD in the sound.
Triodes seem to do this very well. Drive a negative-going signal into the grid, and the heavy curvature of the transfer characteristic near cut-off gently squashes the (positive-going) output signal. Drive a positive-going signal into the grid, and the soft saturation characteristic of the triode gently squashes the negative-going output signal. (Grid current flow at the input might add to the effect.)
A simple JFET source-follower does something loosely similar, though manufacturers are bent on making FETS into bettery switching devices, by sharpening cut-off and saturation, which is exactly the opposite of what we want for gentle and gradual signal peak-squashing.
I certainly didn't invent the concept of "clean" guitar tone still containing audible amounts (several percent) low-order THD. But I wonder if awareness of this is being gradually forgotten, as people who knew both tubes and audio engineering have shuffled off to their final gig in the sky, and those who remain grew up with op-amps and DSP chips?
I fit that category, actually, as I grew up with solid-state electronics, and therefore had no clue what I was missing, or why I sounded so bad, when I began trying to play electric guitar in my twenties, through solid-state guitar amps I designed and built in droves. All of which incorporated what I knew about Hi-Fi solid state amp design, i.e. lots of NFB, low THD, wide, flat frequency response, et cetera.
On the forgetting of old knowledge, there is a parallel with the Theremin. Leon Theremin's prototype instrument, incredibly, dates from 1920, when tubes were the only amplifying devices in existence. Surviving audio clips of old tube Theremins have a rich timbre that is entirely lacking in some of the newer solid-state versions.
Along those lines, I remember horror-stories told by shocked piano teachers in the 1990s, of students who had grown up hearing the bright, ringing "piano" sounds from the Yamaha DX7 electronic keyboard, and who then complained that actual acoustic pianos sounded wrong and fake!
Obligatory Doogie Howser intro music clip, full of bright, ringing DX7 "piano" sounds:
Will we one day have DSP guitar amps emulating the horrid sounds of an overdriven Fender Frontman, in the same way we now have electronic keyboards emulating the cheesy ringing tones of the DX7's "piano" patch?
While we're waxing philosophical, I can't help but notice that e-guitar timbre in popular music went from very, very clean (Les Paul in 1950's) to progressively more and more heavily distorted with every passing decade, from acoustic blues to electric blues to blues-rock to rock to metal. Eventually e-guitars in some popular music sounded like power tools grinding on a tin roof to me.
Example: Evanescence, circa 2005. The grinding noises in the background starting about 40 seconds into the clip are barely recognizable as electric guitars:
Having hit that limit, timbres going in the opposite direction, towards "cleaner" and less distorted sounds, seem to have begun to gain in popularity, for instance in shoegaze music. There's often some sort of distortion buried under the layers of delay and reverb and modulation, but it sounds like crude clipping-diode distortion, not "tubey" or "bluesy" distortion.
More recently, e-guitar seems to be fading fast from popular music, while plugged-in acoustic guitars are still popular. As Thoglette says, acoustic guitar usually involves "Hi-Fi clean" amplification.
Have we come full circle, when it comes to guitar sounds? All the way from squeaky-clean to power-too-grinding, back to squeaky-clean?
-Gnobuddy
You said yourself that the air weighs 'only' about 1.2kg per m^3. That's easily 50kg or more set into motion in a living room. And it's extremely springy. But whatever. Keep thinking that it's detached from the speaker, as if one end of a string can keep itself under tension -- you're confusing resonance with reverb and a diffuse field, which only starts at a higher frequency.
Let me know if you ever actually do that experiment.
But only a very small fraction of that 50kg moving mass couples into the other speaker, hence the “weak” coupling. The rest of it is shaking the room and being heard by listeners. In order to couple ALL of it, the waveguide between the two needs to be symmetrical and can’t leak energy anywhere else.
The results will be completely different with two loudspeakers in opposite corners of the room vs. two loudspeakers strapped together face to face. In the latter case you CAN generate significant power out of the undriven one. Especially if you use horns which further confine the energy. I wonder if you could use it as a direct measurement of efficiency - logic says you could. Might be a good way to quantify box losses, which you don’t necessarily have good numbers for (other than educated guess).
The results will be completely different with two loudspeakers in opposite corners of the room vs. two loudspeakers strapped together face to face. In the latter case you CAN generate significant power out of the undriven one. Especially if you use horns which further confine the energy. I wonder if you could use it as a direct measurement of efficiency - logic says you could. Might be a good way to quantify box losses, which you don’t necessarily have good numbers for (other than educated guess).
Agreed, and with some additional parts you can make them squishier. I was playing around with a couple of JFET stages for a preamp/boost. Putting some local NFB resistors and a single diode gave me this output. Input was 100 mV pp and it's about 20 dB gain. Diodes don't necessarily hard clip or sound harsh, if you're careful.
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Your failure to understand doesn't make it nonsense. Do a slow (30 second) sweep from 20 Hz to 100 Hz, and compare the far-field microphone results w/o MFB.
The trick with resonance is tjat the worse it is, the harder it is to measure because the peaks become extremely narrow.
Your failure to provide evidence for such a nonsense claim speaks for itself ("extraordinary claims require extraordinary evidence"). You didn't provide ANY evidence!
On the other hand, I have serviced several different models of Philips MFB monitors, measured them (I have Audiomatica CLIO) and enjoyed both excellent measurements and excellent sound.
Your turn...
On the other hand, I have serviced several different models of Philips MFB monitors, measured them (I have Audiomatica CLIO) and enjoyed both excellent measurements and excellent sound.
Your turn...
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There's nothing extraordinary about springy materials tending to resonate at certain frequencies. Well, there kind of is, seeing as it's the basis for how tones are made, but nearly everyone here seems to be dismissing musicality and the evidence of their own ears as inferior to some pet theory of how they think physics works.
Brute force control of cone position with no regard or upper limit to the force required to achieve that side-goal leaves the system wide open to misbehave. The crux of the problem is what happens at resonance. Not the normal ~90dB efficiency at the 'tight' frequencies, but the ~100dB+ of the occasional long notes that seem to ring "because of the room". Higher efficiency goes together with the cone having more leverage over the air, basically 'throwing' it like moving your hand 5cm to 'throw' the other end of a slinky spring 20cm. (Basic horn theory. It's much, much more than just narrowing the dispersion angle.) When that happens, the residual SPL added when the MFB applies more force to counteract the 'errors' introduced by the nearfield SPL at resonance starts to add up.
Can you see how the polarity of the feedback is positive? The MFB speaker actively adds more SPL in response to fractional errors caused by already-high SPL.