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I have owned both low and high efficiency speakers. And I strongly prefer high efficiency speakers with horns. But what makes a woofer more or less sensitive (I am using the terms interchangeably) Is it the strength of the magnet? size of the voice coil? other electro magnetic factors?
I assume that like most things, design and construction involve a number of compromises. Are there any downside tradeoffs that come with high efficiency?
Follow Ups:
Sorry, I could not resist 💩
Hi
I think most of the posts have touched on parts of it.
Efficiency is the issue or how much power radiates how much sound.
A loudspeaker radiates sound power into a resistive load that is the air.
There is a radiation resistance curve that is a 6dB/oct slope up to a frequency where the radiator is 1 WL in circumference and known as "K=1"
This means that above K=1, there is no gain in efficiency making the radiator or Horn larger (although doing so causes the pattern to narrow).
In the olden days, they used to say a 30Hz bass horn had to be 10ft in diameter to work. Well a look at bass horns in commercial sound at work shows that's ridiculous BUT IT IS based on the fact that for 30Hz, K=1 is about 10ft in diameter in full space.
So the radiator area is a key part of all this as that set's where you are on the radiation resistance curve. If you place two identical subwoofers side by side, you get +6dB (4x) the amount of bass energy as you have doubled the radiating area. This gain from mutual coupling goes away when the sources are more than 1/3 Wavelength apart.
One can imagine that as the frequency goes up, the acoustic size of the source doubles each octave you go up and that means with the same velocity at all frequencies, the response will climb +6dB /oct, called a velocity response. IF one had a horn which presents a resistive load to the radiator, THAT is the response one would want for the driver.
On the other hand, to make a direct radiator that is acoustically small "flat", one needs to have an acceleration response instead, rolling velocity off 6dB /oct.
How? Go to the link below, scroll down to "electrical subsystem".
Your woofer / cone driver has a series R and (small) L, both are properties of the gauge and length of wire and it's winding into a coil.
Those are in series with the R,L C parallel circuit in the mechanical mobility analogue lower down. This is what the wire in the magnetic field looks like being free to move, attached to a mass C and spring (L the suspension). IT is the Velocity of the radiator / motor that face the radiation resistance.
You can see the parallel RLC tank circuit is where the sealed box impedance peak comes from and at 0Hz or DC, you can see the series R with an ohm meter. Above resonance, the parallel C dominates and now one has a series R, small L driving
Now the ugly wall of science creeps in, Hoffman's iron law.
The upshot is for flat response that no matter what driver parameters you have, even made with unobtainable , your maximum possible efficiency is set by the box volume and -3dB low corner F.
Lower the corner 1 octave for the same box size and the maximum possible efficiency drops 9dB.
Horns have an upper efficiency limit too although the directivity (DI) can cause a large increase in on axis sensitivity that looks like efficiency in front.
Tom Danley
Technical Accuracy and sanity rules. I fear this informational nugget my be lost on some of the lurkers heres. Great explanation about mutual coupling, which is part of my setup and crossover choice for the LFE part of the bass in full Atmos HT or 2 channel stereo with subs.
Any effects of altitude here?
Atmospheric pressure (mmmg) at sea level being 760 mmmg while here in 7,000 ft. Santa Fe it's about 586 mmmg.
Always wondered if that's why there are so many audiophile companies located in Denver and, IIRC, Stereophile used to be headquartered here?
"Reality cannot exist because it cannot keep up with the lies on the Internet."
Yes actually.
The term Rho C is in many calculations and is the density of air (about 1.21 kg per Cubic Meter) times the sound velocity.
Density or pressure does not change the speed of sound but density does change the acoustic load resistance as you suspected and it is reduced at altitude.
How much ambient pressure changes with altitude is interesting
https://www.omnicalculator.com/physics/air-pressure-at-altitude
I think this would apply Underwater as well.
Yes, the same terms come into play too, Rho / density per cu/mtr and C the speed of sound.
The impedance per area are HUGELY different, a PZT ceramic itself is a pretty good impedance match to water's radiation resistance but is close to useless as a direct radiator in air.
In the levitation sources i made a stack of 4 PZT elements and that drove a resonant titanium transformer and flexural radiator that had a Q of around 300-500 and that made an efficient source. One could levitate a styrofoam ball at 10Watts input about 160dB @ 22KHz.
Tom
sounds like loads of fun outside of music
About 1dB less at 6000 feet.
Does that mean my EdgarHorns are only 105dB speakers at 7000 ft?
Dang!
"Reality cannot exist because it cannot keep up with the lies on the Internet."
Probably. Go high enough and they'll be 0dB.
mean speaker are more or less efficient here?
Do they have to work harder to move the same mass of air as they would at sea level?
Gotta be SOME audio benefit from living where there are so few molecules of oxygen in the air.
"Reality cannot exist because it cannot keep up with the lies on the Internet."
Air density determines the impedance load presented by the air. The higher impedance of denser air creates a more efficient energy transfer. It's the same thing that makes horns work. The higher air load impedance of a throat area smaller than the cone along with the mass of the air column creates a more efficient energy transfer than directly coupling the cone to the air.
Edits: 06/16/23
in a word: acceleration. Most high sensitivity drivers have very powerful magnets with low moving mass. The acceleration will translate to the amplitude of the sound wave. The driver doesn't have to move very much to be loud...even for bass (if loaded appropriately).
I have midbass drivers that are 99dB sensitivity and they almost never visually move. Think < 1mm excursion even with strong low frequency content. The magnet is 1.9 tesla and the moving mass only 7 grams.
A compression driver is even more extreme with magnet strength often > 2T and moving mass in the mg range. This means sensitivities in the 105-115dB range (with horn). Acceleration is extreme and therefore the smallest inputs yield high output.
in a word: acceleration.
The moving mass of the 2.5 micron membrane of a Sound Lab full range electrostat is but a fraction of the air load that it is moving and thus can most certainly accelerate .
They are not, however, high efficiency rated at merely 89 db/1W/1M.
Acceleration depends on the field strength as well, not just the moving mass, and it is clear from the lowish sensitivity that it doesn't accelerate that membrane as strongly as a compression driver or other similar high magnetic strength low moving mass diaphragms. What is the total mass of the whole film being moved? It is certainly more than 7 grams.
What is the total mass of the whole film being moved? It is certainly more than 7 grams.
An industry calculator says 7' x 42" x 2.5 microns is ~5.3 gm.
The mass I quoted includes the mass of the air...the driver itself is 3.7 grams. I am guessing the air mass given that large area is a wee bit more??
You forgot to add that they are SEVERELY output limited if you are trying to reproduce the IMPACT of DRUMS which require another 50 Db of output to realize.
This Forum is HIGH EFFICIENCY, so why is your post about Electrostatic Loudspeakers???
This is Horns and High BL drivers territory.
Kind of like talking about boats or planes at a Car Show, aye???
Stats are cool and have in common with horns their extreme clarity. Some years ago when I moved from a large house (in which I had several Altec and JBL systems) to a high rise apartment on the Lake in Chicago I dehorned and bought Martin Logan Vista electrostats and the only thing they gave up to my horn systems was dynamics and I couldn't use extreme dynamics anyway. I now use 604Es in an apartment, not for their dynamics but their tone and clarity.
Edits: 06/10/23
so why is your post about Electrostatic Loudspeakers???
Only too happy to repeat. :)
Brad needs more than "a word" to define high-eff speakers. The one he chose is not unique to them.
Sorry my word is fine...it's your understanding about what is acceleration that is the issue.
If we consider the equation F=ma and it is the force that makes the amplitude then for a given mass, the acceleration determines the force.
You made a comment about how light the membrane is relative to the column of air it pushes...this is the moving mass but says nothing about the acceleration part of the equation.
So, even if the electrostatic membrane is the lightest moving mass in existence (not sure if that is true or not) the fact that the sensitivity is only 89dB (not that bad really but not what I would call high sensitivity) tells me that the electrostatic drive cannot generate that much force and therefore has a lowish acceleration compared to my Supravox drivers or my compression drivers.
"It is not a very long post, but if you are really busy, here is a TL;DR summary:
Displacement in speaker diaphragm does not generate the pressure wave in air that is the sound we hear. Acceleration does.
What causes the diaphragm to accelerate (and therefore the air in front of it to pressurize) is force. A lower acceleration only results in lesser sound pressure level (SPL), i.e. less loud, not how quickly it appears, i.e. the lack of speed.
The speed at which we can modulate "force" is unrelated to the mass of the diaphragm.
I'll expand a little further on Purifi's blog post, since someone will inevitability ignore the last point above and will insist that acceleration is force divided by mass, and therefore lower mass gives higher acceleration. So how do we find how much acceleration we need?
The late Siegfried Linkwitz (RIP) gave us a very handy formula to predict the free field SPL generated by a speaker driver, given its size, diaphragm travel, and frequency. [Link, see the box "Theory Behind the Nomographs"] It is:
SPL = 94.3 + 20 log10(x) + 40 log10(f) + 40 log10(d) - 20 log10(r)
where: x is the peak-to-peak diaphragm travel in meters,
f is frequency in Hz,
d is the effective diameter of the diaphragm in meters (d = sqrt(4 * Sd / pi), with Sd = effective area in m^2)
r is the listening distance in meters
Now, say we want to generate the same SPL at two different frequencies, f1 and f2, what will the diaphragm travels (x1 and x2) be?
SPL1 = 94.3 + 20 log10(x1) + 40 log10(f1) + 40 log10(d) - 20 log10(r)
and
SPL2 = 94.3 + 20 log10(x2) + 40 log10(f2) + 40 log10(d) - 20 log10(r)
Since we want SPL1 = SPL2 ,and "d" and "r" remain the same, we have:
20 log10(x1) + 40 log10(f1) = 20 log10(x2) + 40 log10(f2)
or
log10(x1) + 2 log10(f1) = log10(x2) + 2 log10(f2)
or
x1 * f1^2 = x2 * f2^2
or
x2 = x1 * f1^2/f2^2
So, the amount of travel the diaphragm needs to produce the same SPL in inversely proportional to the ratio of the frequencies squared (i.e. with the same diaphragm travel, SPL goes up/down by 12 dB/octave).
How about acceleration? Well, given the displacement amplitude x, acceleration = x * (2*pi*frequency)^2. Which means acceleration goes up by frequency squared. Since, for the same SPL:
x2 = x1 * f1^2 / f2^2
and
a2 = x2 * (2*pi * f2)^2
= x1 * (f1^2 / f2^2) * (2*pi * f2)^2
= x1 * (2*pi * f1)^2
= a1
The acceleration amplitudes are the same! And therefore forces. Amazingly we need the same force amplitude to produce the same SPL regardless of frequency. Of course, the rate of fluctuation of force is higher with higher frequencies, but the force magnitude is independent of frequency. We need to wiggle the diaphragm more frequently, but that is completely countered by the fact that we need to wiggle it less far.
There are plenty of other reasons why a woofer is not suitable to produce treble. Mass of the diaphragm ain't one."
that any of Siggy's designs are considered "high-eff"?
Is that why the LX521.4 "requires between 8 to 10 power amplifiers of 50 W to 200 W"?
Mass of the diaphragm ain't one.
More than the diaphragm moves. ;)
The LX 521.4 is an open baffle design to remove "the box resonances" and their phase issues.
More importantly, as a SYSTEM, that provides an AMAZING 3-Dimensional presentation of the Music in Space, where the Speakers themselves simply "Disappear" Acoustically speaking.
Their Figure 8 pattern inherently decouples the speakers from the room by Side Cancellations which "throw away" driver Efficiency to achieve all the goals of that type of reproduction.
Having owned 3 pairs of Caver Amazing speakers and helping a friend build his version of it (Six 10" drivers and on large Bohlender Graebener Planar Magnetic driver, side by side), the speakers have the unique presentation that is quite remarkable. The antithesis of my preferred All Horn systems.
Nevermind that he's using JC-1 Plus class A/AB power amps at 1 Horsepower Each to drive them, they just do sound Terrific, at a HIGH PRICE, for the hifi.
I only mention this in CONTRAST to the Horns People here, including me. IMHO, this type of system is not for the Weak Wallet types like me and other Klipscheads from that forum.
Most certainly not in the high-eff family.Never took a cotton to traditional Altecs/K-Horns but would like to hear one of Tom's synergy horns where all frequencies radiate from the same "mouth". In some respects they are like my Sound Labs in that they have controlled directivity and can be used in horizontal or vertical arrays since they are full range designs.
I discovered as a teen that I am particularly sensitive to driver blending issues and have no need for 100 db output. I suspect that music preferences play a role in determining one's optimal speaker approach.
Edits: 06/13/23
I have 3 Danley SH-50's in my full Atmos HT as front RCL channel for a 7.4.2 setup in my tiny Living Room. Also have Tom's prototype #3 of his companion TH-50 tapped horn Subwoofer that literally scares strangers with the right movie LFE.At the touch of a button on my Yamaha Pre Pro, I have 2.2 for Music instantly.
I bought a used pair of his Unity Summation Aperture TD-1 horns. I drove all the way from Indianapolis (at the time) to Buffalo, NY to get them from a DJ who had an extra pair. They easily fit in the trunk of my Toyota, at the time. I was blown away by the sound of those little horns, even without a sub.
So, sound unheard, I upgraded to a pair of SH-50s Brand New, in 2014, and a Hypex NC-400 Kit, also unheard. This combination makes the Chesky Drum Recording sound like the Drums are literally in my living room as certified by a few drummer friends who shook their heads in disbelief. This done in 2.2. Also the Center Phantom image is so sharp on a vocalist, that my friend have to stick their heads in the middle one to know it's not even on. Lots of FUN as the voice sounds like it's coming directly from the center of my big 4K TV with just Land R on. LOL
Now THIS is true HiFi!! Full Dynamics, driven by TI Chip amps, while I reduce the gain on my Hypexes.
My small space means I have to have the horns in corners, which boosts the bass, so I have to EQ them down. This way they perform way beyond the Spec , as flat to 40 Hz than crossed to the subs. I will never sell them as they are a lifetime speaker for me after owning every horn speaker imaginable, several DOZENS by now, and counting.
Edits: 06/13/23 06/13/23
more simply:Acceleration = Force/Mass
Same force, high mass → low acceleration = low SPL
Same force, low mass → high acceleration = high SPLThe relationship (for first order effects) is independent of frequency.
Or if you keep mass constant and increase Force then you get higher acceleration = High SPL
It is the acceleration that creates the pressure wave, not displacement and not velocity the rate of change of velocity (acceleration) creates the pressure wave and it's amplitude (SPL).So, in a word...acceleration defines sensitivity...how you get that acceleration (low mass, high magnetic strength etc.) is the balance played in driver design.
Edits: 06/12/23
"Acceleration is a red herring."
Which chums?
Who is Brad? (like we are supposed to know your friends on a public forum?).
and a 100 db/watt at 1 meter or calculated back at 10 meters in the case of a big E-stat is good enough NOT to need more words to define Sensitivity/Efficiency.
89 'aint cuttin it!
Who is Brad?
Sorry, he's the Swiss based chemist known here as "morricab".
89 'aint cuttin it!
That was my point from the outset. :)
It's basically a Capacitor that moves very little.
for the most linear results. And this one doesn't chop instruments into dissimilar pieces.
The complementary end to your favorite Shoeps or Neumann condenser microphones.
The point of ANY instrument is the generated fundamental and harmonics are all dissimilar sine waves.
I fail to see how your "chopping" analogy is relevant to the FACT that ANY motion reduction in ANY transducer reduces the INTERMODULATION Components that are Artificially Generated by a wide bandwidth transducer at, say 100 db Output. Any further reduction requires another transducer, especially below 300 Hz. where higher excursions are a must at 100 db output.
Acceleration is a red herring. Yes, logically a lower mass diaphragm is more easily accelerated than one with more mass. That would be a concern if the electromotive force of a voice coil or electrostatic grids operated with the realm of the speed of sound. But they don't. They operate within the realm of the speed of light. Making a diaphragm vibrate at 20kHz over a distance of less than a millimeter is no chore for an electron wave that travels some 270,000 km/s.
One advantage of the low mass of an electrostat diaphragm is it stores less kinetic energy to be released in the form of distortion. The large surface area reduces the excursion required to reach a given SPL, also reducing distortion. Shorter excursion to reach a given SPL also results in lower acceleration, as the diaphragm doesn't have to move as far within the same time frame, so even if acceleration mattered there's less need for it with an electrostatic.
No, it is not a red herring. Acceleration is the rate of change in velocity and this has a direct impact on the amplitude of the wave generated.
If I have a driver that moves 1mm in 1 millisecond vs. the same driver that moves 1mm in 10 microseconds, which one do you think will have the higher amplitude? The force of compression of air is much greater with traveling the same distance in a much shorter time frame.
A larger surface area will also give a higher sensitivity for a given force as clearly more air is being moved. A large e-stat panel will therefore be able to make a reasonable SPL level without having a particularly strong motor and therefore somewhat slower acceleration.
You can make ribbon speakers with high sensitivity as well...but it takes very powerful magnets to do so to generate the acceleration needed for high sensitivity. Again, back to acceleration.
A diaphragm accelerates as fast as it needs to for the required excursion and frequency. In your example of 1mm in 1 millisecond vs. the same driver that moves 1mm in 10 microseconds they would not reproduce the same frequency. A wave period of 1 millisecond is 1kHz. A wave period of 10 microseconds is 100kHz.Acceleration only enters the picture when comparing two different excursions. A given tone at 1mm excursion would require ten times the acceleration as the same tone at 0.1mm excursion. If the transducer lacked the necessary acceleration to realize the given excursion within the wave period the result would be a lower frequency, not lower sensitivity.
Edits: 06/12/23 06/12/23
"It is not a very long post, but if you are really busy, here is a TL;DR summary:
Displacement in speaker diaphragm does not generate the pressure wave in air that is the sound we hear. Acceleration does.
What causes the diaphragm to accelerate (and therefore the air in front of it to pressurize) is force. A lower acceleration only results in lesser sound pressure level (SPL), i.e. less loud, not how quickly it appears, i.e. the lack of speed.
The speed at which we can modulate "force" is unrelated to the mass of the diaphragm.
I'll expand a little further on Purifi's blog post, since someone will inevitability ignore the last point above and will insist that acceleration is force divided by mass, and therefore lower mass gives higher acceleration. So how do we find how much acceleration we need?
The late Siegfried Linkwitz (RIP) gave us a very handy formula to predict the free field SPL generated by a speaker driver, given its size, diaphragm travel, and frequency. [Link, see the box "Theory Behind the Nomographs"] It is:
SPL = 94.3 + 20 log10(x) + 40 log10(f) + 40 log10(d) - 20 log10(r)
where: x is the peak-to-peak diaphragm travel in meters,
f is frequency in Hz,
d is the effective diameter of the diaphragm in meters (d = sqrt(4 * Sd / pi), with Sd = effective area in m^2)
r is the listening distance in meters
Now, say we want to generate the same SPL at two different frequencies, f1 and f2, what will the diaphragm travels (x1 and x2) be?
SPL1 = 94.3 + 20 log10(x1) + 40 log10(f1) + 40 log10(d) - 20 log10(r)
and
SPL2 = 94.3 + 20 log10(x2) + 40 log10(f2) + 40 log10(d) - 20 log10(r)
Since we want SPL1 = SPL2 ,and "d" and "r" remain the same, we have:
20 log10(x1) + 40 log10(f1) = 20 log10(x2) + 40 log10(f2)
or
log10(x1) + 2 log10(f1) = log10(x2) + 2 log10(f2)
or
x1 * f1^2 = x2 * f2^2
or
x2 = x1 * f1^2/f2^2
So, the amount of travel the diaphragm needs to produce the same SPL in inversely proportional to the ratio of the frequencies squared (i.e. with the same diaphragm travel, SPL goes up/down by 12 dB/octave).
How about acceleration? Well, given the displacement amplitude x, acceleration = x * (2*pi*frequency)^2. Which means acceleration goes up by frequency squared. Since, for the same SPL:
x2 = x1 * f1^2 / f2^2
and
a2 = x2 * (2*pi * f2)^2
= x1 * (f1^2 / f2^2) * (2*pi * f2)^2
= x1 * (2*pi * f1)^2
= a1
The acceleration amplitudes are the same! And therefore forces. Amazingly we need the same force amplitude to produce the same SPL regardless of frequency. Of course, the rate of fluctuation of force is higher with higher frequencies, but the force magnitude is independent of frequency. We need to wiggle the diaphragm more frequently, but that is completely countered by the fact that we need to wiggle it less far.
There are plenty of other reasons why a woofer is not suitable to produce treble. Mass of the diaphragm ain't one."
All this math very few people will bother to try and understand is a waste of time and effort. Modern hifi is much simpler than that with the tools that do all the math for you.
Go listen to some music on whatever you have and be happy.
more simply:Acceleration = Force/Mass
Same force, high mass → low acceleration = low SPL
Same force, low mass → high acceleration = high SPLThe relationship (for first order effects) is independent of frequency.
Or High force, same mass - high acceleration = high SPL
Low force, same mass - low acceleration = low SPL
Edits: 06/12/23
Exactly! Direct radiators (K < <1), to have flat response have an acceleration controlled response.
I think the hard part to get is how the motor works, it's a small series inductance and a DC resistance in series with the loss less motor BL, that BL can also be expressed as Newtons of force per Amp of current but it is also a generator who's voltage proportional to BL and the motion velocity and called back EMF. The higher the BL, the greater the back EMF, they are hand in hand.
Doesn't this high BL woofer then, as a generator of back EMF benefit from a class D amp like a Hypex with strong feedback and relative load independence down to 2 ohms? IOW, an amp with strong control over the motor vs. a wimpy Single Ended Tube amp with a few watts on a good day?
This is complicated as there is more involved than just motor strength.
What you have is a mass and spring (moving mass and driver plus box compliance or spring) Those two reactance's have a resonance where the two reactance's are equal but opposite and cancel out. AS you approach resonance, you see the parallel L and C's effects cancel out and at Fb resonance the only load you see in the impedance peak is the Qm mechanical losses and a tiny amount of radiation resistance (big ohms).
The motor is connected to this moving stuff and so is the radiator.
The bigger and more massive the moving system, the larger the BL has to be to have the same amount of "control" on the moving system.
The amplifier (in modern times) is assumed to be a low impedance voltage source. It used to be a selling point that the amplifier's output impedance was tiny with respect to the load and the ratio expressed as "damping factor".
In loudspeaker land, the driver has a DC resistance (Rdc) in series with the motor so the amount of "control" is limited by that Rdc (BL and Rdc controlling motor part of the driver Qe). The amplifier appears to be a small resistance in series also but if the load were 4 ohms and the DF 100, the amplifier looks like 4/100 Ohms. Around 20 or 30 DF is where making it larger has very little/ less and less effect.
A series resistance that is significant (say an Ohm) begins to modulate the normal frequency response, where ever there is a low impedance, there is a new small dip or depression added and so the shape of the response curve is altered by the addition of a version of the shape of the impedance curve.
Morricab brings up an obscure but real issue.
The assumption is when you add negative feed back to make a closed loop, you lower the output impedance accordingly as well as lower the distortion. That is true to the degree there is open loop gain left to do those things at what ever frequency it is.
If one connects a capacitance across an amplifiers output and drives it with an impossible to follow signal like a square wave, theoretically what one would see is the square wave rounded off in an R/C low pass filter (amplifier output R and load C).
In reality, normally what one gets is a ringing square wave showing not a simple but a more complex and reactive source, showing an amplifier over shooting and damping often with a definable Q.
In my own fooling around, what found was if one made an amplifier with a low output Z without -feed back, it did behave like an R/C.
I think what one can say is in a closed loop reactive loads can use up the open loop gain that normally gives the good measurements.
How big is this stuff?? I don't know but at least with an amplifier, it's possible to set up a null test where one can listen to the difference between the input signal and output signal, much harder with loudspeakers but the generation loss test works.
Tom
Matti Otala showed that back EMF has more impact on the sound quality of high feedback amps as that back EMF is pushed through the feedback loop and re-amplified. A SET without feedback will just dissipate the back EMF as heat in output transformer as there is no backwards path through the amp without a feedback loop. A high sensitivity driver with a low Qts doesn't need or want a high damping factor amp...- my 99dB Supravox drivers with a 1.9T Alnico magnet and 7 gram MMS and Qts of 0.21 love SET above all else.
Hypex Class D is a different animal opposite of an SET amp. Not saying it's better, and I know not why it takes full, tight, control of all the voice coils regardless of the size of Magnets.
All I have practiced since 1976, is what Saul Marantz told me when I talked to him at a Detroit Audio Show. Solid State on the Bass, Tubes on the mids and highs. He was President of Dahlquist then and sold his name to Superscope for a tidy sum, I imagine.
I have found it is better for coherence to have the same type of amp on each driver. You can hear that different amp topologies are at play when done like you suggest.
> You can hear that different amp topologies
You shouldn't, since where they differ is mainly in the midrange, where subs don't operate. SS where more power was needed and tubes where it wasn't was a common approach in the early days of SS. Things have changed greatly since then. I doubt anyone finds a Bryston or Powersoft to be wanting compared to tubes today, other than where confirmation bias determines the result.
Oh but you can hear the difference easily. I was at a friend's yesterday and he has two amps, one SS (Plinius SA103, which is a Class A PP SS amp) and one SET (Ayon Helios, which has 2 x 6C33C tubes per channel in Parallel SET). We listened to both digital (Ayon Stratos) and Analog (Transrotor + Transfiguration Proteus and Chinook phono).
We started with the Plinius and the sound was, not too bad. Decent tone, ok dimensionality (both soundstage and imaging) and tight powerful bass. Two things stood out; 1) the bass sometimes seemed a bit dominant in the overall sound signature and 2) vocals were quite present but the rest of the instruments in the mix "sat back" a bit.
Then we switched to the Ayon Helios...the difference was not subtle. Much more accurate tone color of instruments, better dynamics, instruments that were there but not recognized popped out. Dimensionality in all directions and imaging 3d improved drastically. Finally, and importantly, the bass was now less prominent but seamlessly integrated into the whole, where before it stuck out and called attention to itself.
I have heard this effect in the past as well. I once took a KR Audio VA350i to a guy running a Gamut D200 amp. With the Gamut the speakers sounded like a collection of drivers, each kind of doing it's thing. With the KR it sounded like they were all working together for the whole sound.
So, in essence I think you are completely wrong to assume a Bryston or Powersoft will be difficult to tell from a good tube amp...nothing could be further from the truth...it is blindingly obvious.
Read my post again: "where they differ is mainly in the midrange, where subs don't operate."
As for a Bryston or Powersoft not equaling or bettering a good tube amp, show us the evidence. That means measurements and...shudder,shudder,shudder..double blind testing.
99% of Subjective words without data to back them up deserve the Paul Klipsch Yellow Button.
"Confirmation Bias" and no AB/X test box (I was there for the unveiling of that rarely utilized device when I was an AES member at a local chapter meeting in Michigan).
Also no measurements to back up comments from 90% of people on the web (I could be understating it in their favor).
Knowing I am with fellow cheapskates here (a distinguished group I might add)..................who's gonna spend a Kilobuck on a multitap autoformer with a rotary control for only ONE friggin driver????
Go active crosovers with 2, 4, 6, 8, 10 outputs, since we are talking high efficiency, you only need TI 3255 chip amps and derivatives to do the job. They are available RCA or Balanced XLR instead of buy 1 Horsepower Class A/AB amps like a friend just did Times 2 because they are monoblocs.
You specifically ask about drivers, so ignoring the enclosure design (very important in its own right), the main influence on sensitivity of a driver is its magnet. A small current passing through the voice coil will move further and faster if it's contained within a powerful magnet. The closer to the centre of magnetic influence the better.
The further and faster the coil moves, the greater its sensitivity as the resultant sound vibrations that reach your ears will be louder for a given coil current.
Someone will almost certainly shoot down this explanation, but there we go!
The further the coil, and therefore the cone, moves the louder the sound. The faster it moves the higher the frequency. Neither has much to do with driver sensitivity. The strength of the magnet and the size of the gap between it and the coil do.
Yes, that's what I said though you misinterpreted the use of the word "faster". It's the faster the coil accelerates that has a bearing on sensitivity and sound quality, not the signal frequency. It's the strength of the magnet and its effect on the coil that's the major factor.
The coil will move further and faster (nothing to do with signal frequency) that determines the resultant volume and this has everything to do with sensitivity. The louder the sound for a given current passing through the coil, the higher the sensitivity.
Current doesn't affect excursion. Voltage swing does. With a given voltage swing the cone/coil of a higher sensitivity driver will move further than a lower sensitivity driver. By the same token a lower sensitivity driver requires higher voltage swing to realize the same excursion as a higher sensitivity driver. It's as simple as that. Bringing coil acceleration into the picture just clouds the issue, as it's a different question entirely.
Bill- You're splitting hairs. An increase in either voltage or current will increase the power being sent to the speakers, but this is academic.
Truth is that the more powerful the magnet and the closer the coil is to the magnetic force, the greater the sensitivity of the driver. So, it's all done by magnets as my tongue-in-cheek revised Title implies.
Splitting hairs is what transducer engineers and loudspeaker designers do. It's how we make our living. Current is moot. What causes a cone to move is voltage. Unless you have a transconductance amplifier you can't increase current to realize more output, you increase voltage. Magnet size affects sensitivity, but so does coil gap, Mms, Qts, Cms, Le and more. So yes, it can be done with magnets, but by no means is it all done with magnets. For every driver with a given size magnet there's a lower sensitivity driver with a larger magnet and a higher sensitivity driver with a smaller magnet.
> Splitting hairs is what transducer engineers and loudspeaker designers doI don't think the OP wanted hairs split
> Magnet size affects sensitivity, but so does coil gap, Mms, Qts, Cms, Le and more.
Exactly what I was saying, but in words that may make more sense to a non-speaker designer
Let's leave it, we both know what makes a sensitive driver and hopefully the OP has had his question answered!
Edits: 05/30/23
High sensitivity (high efficiency is the wrong term, but I doubt the forum wants to change the name) comes from a combination of factors, including high magnet strength and low moving mass. There are at least fourteen Thiele/Small specs that affect the result. Qes and Mms are only two of them.
The root issue is the radiation resistance, that is the "acoustic load" a radiator feels and is the loudspeakers sound power.
This "load" is tiny and for a single direct radiator is usually so small it is hard to identify in the driver's impedance curve.For example, in free space, 1 Acoustic Watt radiated from a tiny source (no directivity) produces 109dB at 1 Meter or 112dB at 1 Meter in half space (on the ground). So one can figure that a speaker that produced 92dB at 1 W 1m is approximately 1% efficient and 82dB ~ 1/10% efficiency.
As might be apparent most of the energy going to the driver is used simply moving the mass back and forth, the radiation is a very small part of the load. In fact it is the mass that makes a woofer have flat response, it has an acceleration controlled behavior. The weirdest part is that if one added additional mass to the vc (so it didn't effect the cone stiffness etc), the hf response does not change at all but the low F corner and efficiency both go down..
Hoffman's iron law and several other unpleasant realities make low frequencies the hardest to produce / rather more correctly the same rules apply except the size of the wavelength is much larger and so then is the device. Consider at 20KHz the wavelength is 1/1000 the size it is at 20Hz.
For a given box volume, with the perfect driver for the job, losses about 9dB sensitivity per octave as you lower the low corner F.Horns can both confine the radiation pattern angle, making the on axis sensitivity higher AND they can also raise the actual efficiency of the system by coupling the load at the big end to the driver at the small end.
Compression drivers appear to be efficient and some are in the mid range but above say 5Khz, most depend on the directivity to get high numbers.Reality is the actual acoustic power from the compression driver is rolling off starting at a couple KHz and above. On a CD horn (a horn that doesn't narrow down in angle up high in Frequency), the driver's true power response is revealed and so us usually compensated with a 6dB / oct rising EQ to make "flat" response.
Sort of an overview, hope that helps
Tom
Hi Bill
Oops i intended this to be a reply to the OP, not you.
Have a good weekend
Edits: 05/28/23
If you haven't already seen it, you might enjoy the link below. Harry Olson (at RCA Labs back in the 1950's) did some amazing theoretical work on acoustics, speakers, and the like. You don't have to wade into the mathematics if you don't feel like it - his text alone is pretty illuminating.
Why horns have higher sensitivity, and why direct radiators have lower sensitivity, is simple: impedance. For efficient energy transfer you need a low impedance source and a high impedance load. That's the case with electronic devices, it's the case with speakers. The problem is that the load that speakers deal with is air, and air has very low impedance. If we could use very low impedance drivers, say 1/100 ohm, we could realize far better sensitivity, but at the cost of current demands that our amplifier technology cannot even approach. A horn is an impedance transformer, just like a transformer is in an electronic circuit. It better matches the high impedance of the driver source to the low impedance of the air load. It's not all that different than how the output transformers of tube amps match the higher impedance tube sources to the lower impedance speaker loads.
He has to be the most prolific inventor in audio there ever was and as the late Don Davis said " the ancients are stealing our inventions".
I used some of his approach to the explanation in the chapter on loudspeakers in the book below. I tried to make it a visual analogy explanation as previous issues were heavy on math but shallow on explanations someone unfamiliar would understand.
https://www.routledge.com/Handbook-for-Sound-Engineers/Ballou/p/book/9780415842938
In response to the quasi logarithmic treatment of sound radiation and efficiency, what is ONE acoustic watt?All the while knowing that very FEW home loudspeakers can even produce that much sound, except for commercial ones in the home.
Edits: 05/29/23 05/29/23 05/29/23
Well, if one does work, energy is expended and that is the energy radiated away from a loudspeaker.
Of politicians in England my old boss Roy (an acoustician from England) once said something like "if you took all the energy in sound of politicians arguing for a week, you could heat a cup of tea".
His meaning was our ears are very sensitive and the amount of actual energy in normal sound is very small.
So with 1 horsepower being 746 Watts and 1 Watt being 1/746 Horsepower AND that is subjectively LOUD, puts the ears sensitivity in perspective, also at the big end, what's needed to radiate multiple horsepower in stadium or outdoor event.
Great answer. All that remains is Distance and Size of Indoors or Outdoors. I'm sure the rock quarry near your house is really quiet today!BTW, my good friend retired Architect/Builder is a real speaker craftsman. He just bought Parasound JC-1+ monobloc power amps that are Class A/AB (after the first 30 Watts) that have 1 Horsepower per channel.
With 140 Amps of current in the Specs, I asked him if he wanted to use them as Arc Welders too!
Space heaters driving inefficient Planar Magnetics/tall array of Open Baffle woofers require lots of air space, especially in the summer. But they are near open windows most days now. LOL.
Meanwhile, I'm driving all my horns with TI 3255 chip amp derivatives. What a World, this audio stuff, all for our favorite musical illusions.
Edits: 05/29/23
nt
Open up your mind, in pours the trash. - Meat Puppets, 1987
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