caninus80 said:
I was trying single raal 14-150 with altec biflex and raal is too fast. The blend is not seamless
I have to look for another drivers.
Some advice from DIY friends?
Lynn sorry for disturb Your thraed...
And sorry for me english.
Best
C
The biggest difference between ribbons, planars, electrostatics, etc vs a standard VC driver driver is the inductance. The inductance and the related issues due to inductance are a much more serious issue with the "slow" sound than is the mass of the driver alone. In many cases, drivers can play up higher in frequency where they are actually counting on cone breakup to contribute to the output level. They aren't however electrically flat or even close to flat in the range they need to be used.
Lynn mentions:
A lot of this can be accounted for based on the modulation of the flux field due to back EMF created by the coil moving through the gap. In a perfect unchanging flux field, as the coil travels in a positive direction and then reverses, the coil would simply change direction and move back the other way immediately. However, the field created by the energized coil modulates the field in the gap as it moves in a given direction. If the current was removed from the coil, it would take a brief period of time for the flux field to return to normal. As the current is reversed, this modulation of the flux field not only has to return to it's unmodulated state but move past that into it's new modulated state as the coil goes in another direction. This is a serious cause in both distortion and visible artifacts in an impulse response, CSD, etc. One that can be measured by simply building two identical drivers, one with a standard motor and one adding a proper shorting ring to short the currents created as the coil moves through the gap. You will see dramatic differences with the shorting ring. This, among other benefits, is why we use the full copper sleeve on the pole of nearly all of our drivers.
People often bring up the "acceleration factor" that a driver is slow or fast, etc. The acceleration factor has often been described as BL/mass which is not the case in a loudspeaker driver. It is true that force/mass = acceleration, but BL alone is not equal to force in a driver. In reality, the speed is simply proportional to the frequency being played. If you increase frequency by one octave, your speed is doubled for the driver to complete the waveform. Here is a good white paper that Nick at Lambda Acoustics had written back in 1999 explaining this better.
http://web.archive.org/web/20010810141852/lambdacoustics.com/library/whitepapers/bl_mms.htm
John
Actually "BL * VC current" does not mean anything.
Its the area "Sd" that produces SPL - when moved.
To me - impedance does not mean anything - regarding "speed" – simply apply more voltage if you like to have more current.
VC impedance is just another parameter to balance other effects affecting frequency response if desired.
There isn't any definition of speed so far – it's more or less a subjective term.
I like to use "speed" the way to calculate BL for a given Sd - giving a rough gues how "fast" the air pressure ( = SPL ! ) may oscillate between positive and negative values for a certain voice coil current.
This by no way is kind of physiks just a simplified picture for easy understanding.
Basically very similar to magnetars point of view assuming that nothing less than 103 dB in efficiency is worth a try
Greetings
Michael
Its the area "Sd" that produces SPL - when moved.
To me - impedance does not mean anything - regarding "speed" – simply apply more voltage if you like to have more current.
VC impedance is just another parameter to balance other effects affecting frequency response if desired.
There isn't any definition of speed so far – it's more or less a subjective term.
I like to use "speed" the way to calculate BL for a given Sd - giving a rough gues how "fast" the air pressure ( = SPL ! ) may oscillate between positive and negative values for a certain voice coil current.
This by no way is kind of physiks just a simplified picture for easy understanding.
Basically very similar to magnetars point of view assuming that nothing less than 103 dB in efficiency is worth a try
Greetings
Michael
John_E_Janowitz said:
I disagree with much of this. (..though not about reducing back emf and engineering the driver to properly halt its progression before it leaves its gap. Note however that shorting rings (emphasis on plural) is considerably less effective than a counter wound coil a' la 18 Sound "Active Impedance".)
The "slow sound" is principally NOT due to inductance levels, but rather a lack of efficiency and increased moving mass. There are any number of drivers that bear this out.
Inductance will of course have an impact on high freq. loss - and therefor will be physically slower in its ability to oscillate, but that is NOT related to the subjective sense of "slow", it is however *related* to the subjective and objective loss of high freq.s.
Now the resistance portion of the VC (which is literally "inter-wound" with inductance), DOES effect the subjective sense of "slow", but it can also be compensated for by lowering mass and increasing gap strength.
Increased inductance usually does increase harmonic distortion at higher freq.s.
"Acceleration Factor" in the context of loudspeakers is simply an industry term specific to loudspeaker drivers. And yes, it does seem to have a rather good correlation with a driver sounding more or less "slow".
mige0 said:
Problem solved - just use a current source. But then that means you have to design for it from the ground up.
This whole issue of perceived speed, and other similar descriptions, I find frustrating, because the term has no real meaning. That doesn't mean it doesn't exist, just that someone reading here can't be sure what it means, so it could mean anything. That's a prescription for endless unproductive argument.
So let me take a stab, which may be no more right or wrong than any other: It seems at least plausible that it has nothing directly to do with the frequency response of the driver or acceleration of the cone. The simplest explanation (and therefore a good starting point) is that it could mainly be a function of the acoustical properties of the cone - the way it resonates and damps. Lynn said, light cones sound light, heavy cones sound heavy. Maybe the same applies to stiff, soft, metallic, etc.., not acceleration, per se. If it's not acting like a perfect piston -and it never is - you'll hear it. We associate light with fast, so we describe light fairly stiff cones as fast sounding. I'm sure a better, more measureable description could be formed.
Sheldon
Sheldon said:
I think thats part of it - an increase in clarity.
But its also true that it can actually sound *faster* - i.e. peak transients seem to be quicker, more "real". In a speaker that does sound "slower" those dynamic transients seem to lag, not just "blunted" - but sound reproduced latter in time.
Interestingly enough, removing passive crossover parts can in many cases provide a similar sense of "speed". Nor is it simply an increase in freq. extension for a given spl that adds to this sense of "fast".
Boy, I knew things would go into the soup once the words "slow" and "fast" came up. These two subjective terms cause more confusion than any others.
In terms of subjective perception, it is easy enough to equalize two dissimilar systems to measure just the same - but of course we ask "which measurements?" On-axis response, averaged measurements over a frontal arc, power into a sphere? Aside that mess, the equalized systems will still sound different - most noticeably if the diaphragms are dissimilar.
I contend that the voice-coil coupling is loose enough in most drivers there is plenty of room for mischief when the signal from the amplifier stops. Not starts - let's not get all involved in BL factor, cone mass, and all that. Let's look at what happens when it stops.
A driver that "starts up" in microseconds takes much longer to discharge the energy when the electrical input ceases, and slowly releases this energy in many different resonances of long duration and high Q. The mismatch in acoustic impedance between the air-load and the mass of the diaphragm also results in the very little damping from the air - the damping that occurs is mostly the result of inherent mechanical self-damping (conversion into heat). The less the inherent mechanical losses of the diaphragm, the higher the "Q" of the associated resonances, and the longer it takes them to fully discharge their energy.
This "ring-down" is the characteristic signature of the diaphragm itself - the source of my flip remark that "heavy sounds heavy, and light sounds light." Similarly, metal sounds metallic, paper sounds like paper, and plastic sounds like plastic. The inherent material colorations can be attenuated or damped, but I very much doubt they can be eliminated entirely.
Perception makes a difference too - the ear/brain/mind has little to go on with the initial attack, but as time goes on, there is plenty of data available for cross-correlation during the much longer decay phase. What makes this more problematic is the characteristic ring-down sound of the diaphragm interferes with the characteristic sound of the musical instruments and their set of (desirable) colorations.
So I don't think the subjective sensation of "speed" has anything to do with rise times at all. I think it has much more to do with the time signature, or decay properties, of the diaphragm, as well as IM distortion with complex, spectrally dense material.
Looking at different types of diaphragm, a freely suspended ribbon tweeter has a single, one-piece diaphragm & voice coil, but the ribbon also has much lower BL product than other types (only one turn, after all - this results in very loose amplifier coupling). A stretched-film quasi-ribbon has to drag a plastic film along with it, so the problem of separation of voice coil and diaphragm are re-introduced. A dome tweeter and compression-driver dome have separated voice coils and diaphragms, and the voice coil's ability to damp the inherent breakup modes of the dome is very limited. The loosest coupling of all is a cone loudspeaker, where there are several mechanical interfaces between the voice coil and the radiating membrane - even the glues can make a very substantial difference to the sound of the speaker.
(The mechanical path for a cone driver is: VC -> glue -> VC former -> glue -> base of cone -> various losses as the shock wave propagates outward from the base of the cone through the cone -> glue -> surround -> reflections backward into the cone, and so on. The spider and surround create independent sets of reflections and resonances, and both are required for the loudspeaker to operate. Any change in material at any step in the path will change the sound of the driver, which is why re-coning an irreplaceable vintage driver is not a trivial exercise.)
Every time the shockwave from the voice coil traverses a material with a different speed of sound, there are opportunities created for reflections across the speed-of-sound boundaries. All of the materials that are vibrating have inherent sets of resonances, and as mentioned earlier, the voice coil only has limited abilities to damp these vibrations. VC damping is most effective at the very lowest frequencies, where the various mechanical subassemblies have the tightest coupling. As the frequency goes up, the tightness of coupling becomes looser, the first group of resonances appear, and inherent self-damping of the materials becomes more important.
BL product and cone mass dominate the rise time function, but have very little to do with decay performance. That's a whole different set of problems entirely, and one where the materials involved display their characteristic mechanical signatures.
In terms of subjective perception, it is easy enough to equalize two dissimilar systems to measure just the same - but of course we ask "which measurements?" On-axis response, averaged measurements over a frontal arc, power into a sphere? Aside that mess, the equalized systems will still sound different - most noticeably if the diaphragms are dissimilar.
I contend that the voice-coil coupling is loose enough in most drivers there is plenty of room for mischief when the signal from the amplifier stops. Not starts - let's not get all involved in BL factor, cone mass, and all that. Let's look at what happens when it stops.
A driver that "starts up" in microseconds takes much longer to discharge the energy when the electrical input ceases, and slowly releases this energy in many different resonances of long duration and high Q. The mismatch in acoustic impedance between the air-load and the mass of the diaphragm also results in the very little damping from the air - the damping that occurs is mostly the result of inherent mechanical self-damping (conversion into heat). The less the inherent mechanical losses of the diaphragm, the higher the "Q" of the associated resonances, and the longer it takes them to fully discharge their energy.
This "ring-down" is the characteristic signature of the diaphragm itself - the source of my flip remark that "heavy sounds heavy, and light sounds light." Similarly, metal sounds metallic, paper sounds like paper, and plastic sounds like plastic. The inherent material colorations can be attenuated or damped, but I very much doubt they can be eliminated entirely.
Perception makes a difference too - the ear/brain/mind has little to go on with the initial attack, but as time goes on, there is plenty of data available for cross-correlation during the much longer decay phase. What makes this more problematic is the characteristic ring-down sound of the diaphragm interferes with the characteristic sound of the musical instruments and their set of (desirable) colorations.
So I don't think the subjective sensation of "speed" has anything to do with rise times at all. I think it has much more to do with the time signature, or decay properties, of the diaphragm, as well as IM distortion with complex, spectrally dense material.
Looking at different types of diaphragm, a freely suspended ribbon tweeter has a single, one-piece diaphragm & voice coil, but the ribbon also has much lower BL product than other types (only one turn, after all - this results in very loose amplifier coupling). A stretched-film quasi-ribbon has to drag a plastic film along with it, so the problem of separation of voice coil and diaphragm are re-introduced. A dome tweeter and compression-driver dome have separated voice coils and diaphragms, and the voice coil's ability to damp the inherent breakup modes of the dome is very limited. The loosest coupling of all is a cone loudspeaker, where there are several mechanical interfaces between the voice coil and the radiating membrane - even the glues can make a very substantial difference to the sound of the speaker.
(The mechanical path for a cone driver is: VC -> glue -> VC former -> glue -> base of cone -> various losses as the shock wave propagates outward from the base of the cone through the cone -> glue -> surround -> reflections backward into the cone, and so on. The spider and surround create independent sets of reflections and resonances, and both are required for the loudspeaker to operate. Any change in material at any step in the path will change the sound of the driver, which is why re-coning an irreplaceable vintage driver is not a trivial exercise.)
Every time the shockwave from the voice coil traverses a material with a different speed of sound, there are opportunities created for reflections across the speed-of-sound boundaries. All of the materials that are vibrating have inherent sets of resonances, and as mentioned earlier, the voice coil only has limited abilities to damp these vibrations. VC damping is most effective at the very lowest frequencies, where the various mechanical subassemblies have the tightest coupling. As the frequency goes up, the tightness of coupling becomes looser, the first group of resonances appear, and inherent self-damping of the materials becomes more important.
BL product and cone mass dominate the rise time function, but have very little to do with decay performance. That's a whole different set of problems entirely, and one where the materials involved display their characteristic mechanical signatures.
This is probably obvious to all, but the complex mechanical assemblies of loudspeakers have very distinctive sounds when stimulated by an impulse and the initial electrical energy is removed from the system. Big heavy systems with many moving parts sound just like what they are, just as little systems with fewer parts sound like what they are as well.
The driver designer can add mechanical damping in strategic places, thus removing energy from the resonant mechanical parts, but the resonances cannot be reduced to zero. They will remain, attenuated by 5~15 dB (if you're lucky), but still present, and very likely still audible. As mentioned early, thanks to loose, multielement coupling between the voice coil and diaphragm, amplifier damping is really only effective at the lowest frequencies - it's not going to do very much for resonances at 1 kHz and above.
Frequencies 1 kHz and above are where the awkward choices between diaphragm rigidity (lower IM distortion) and increased energy storage from higher-Q resonances play out. The lossier diaphragms (paper or plastic, ma'm?) discharge energy more quickly, but also flex more, thus higher IM distortion. Composites can have the worst characteristics of each, not necessarily the best. I've been hearing claims of "perfect" diaphragms since the early Seventies, so it's a little hard to take any of them seriously.
Once you step out of the marketing world of perfection-if-you-just-pay-enough, all diaphragm materials, along with their suspension systems, have a different series of tradeoffs. It's up to the speaker designer to select a set of tradeoffs that are compatible with the goals of the design.
The driver designer can add mechanical damping in strategic places, thus removing energy from the resonant mechanical parts, but the resonances cannot be reduced to zero. They will remain, attenuated by 5~15 dB (if you're lucky), but still present, and very likely still audible. As mentioned early, thanks to loose, multielement coupling between the voice coil and diaphragm, amplifier damping is really only effective at the lowest frequencies - it's not going to do very much for resonances at 1 kHz and above.
Frequencies 1 kHz and above are where the awkward choices between diaphragm rigidity (lower IM distortion) and increased energy storage from higher-Q resonances play out. The lossier diaphragms (paper or plastic, ma'm?) discharge energy more quickly, but also flex more, thus higher IM distortion. Composites can have the worst characteristics of each, not necessarily the best. I've been hearing claims of "perfect" diaphragms since the early Seventies, so it's a little hard to take any of them seriously.
Once you step out of the marketing world of perfection-if-you-just-pay-enough, all diaphragm materials, along with their suspension systems, have a different series of tradeoffs. It's up to the speaker designer to select a set of tradeoffs that are compatible with the goals of the design.
Lynn Olson said:
This seems most likely to me. Simply tapping lightly on the cone reveals a different signature for different cone materials and weights. This has to be audible. We are used to these sounds in everyday life. Doesn't mean we can't be fooled, but it is generally true that light sounds light, metal sounds like metal, etc.. The dry engineering perspective would want a cone with as close to no ring down, as possible. Even if it were possible, it might not be preferred by many.
Sheldon
Driver sound
Are there any real world measurements anyone has made to determine the audibility of this "material sound" hypothesis causing the perceived differences between various drivers? perhaps a differential between carefully setup driver pairs responding to the same impulse/signal but setup as subtractive such that only the error from the intended sound is revealed?
Any drivers that have been identified as "reference" reproducers displaying a minimum of these colorations?
John L.
Lynn Olson said:
Are there any real world measurements anyone has made to determine the audibility of this "material sound" hypothesis causing the perceived differences between various drivers? perhaps a differential between carefully setup driver pairs responding to the same impulse/signal but setup as subtractive such that only the error from the intended sound is revealed?
Any drivers that have been identified as "reference" reproducers displaying a minimum of these colorations?
John L.
Well, that's what a CSD shows - a mechanically "perfect" driver would just have a narrow wall at the back of the display. Unfortunately, a CSD can be scrambled and rendered unreadable by an unsuppressed floor reflection, but I seem to get usable (not perfect, but good enough for my purposes) results with 20~30 dB of attenuation of the floor bounce.
My web page is a once-over-lightly on what you can expect to see when the impulse response is transformed to a CSD and FR plot. My favorite odd duck is the Eton CSD, shown below. I've never seen this behavior before - the obvious resonance (that long mountain range) is actually frequency-modulated! Remarkable! That shouldn't even be possible, if you stick to lumped-element theory. The real world is always a lot more complicated than our simplistic models - and the deeper you look, with more powerful measurement techniques, the more you find.
The original CSD was D.E.L. Shorter's chopped-sinewave method described in Wireless World in the early Sixties. I wish I had copies of these articles - they were as significant as Theile/Small theory, but were not appreciated in the USA when they came out. (That transatlantic accent, y'know, along with a good dose of NIH in the American speaker industry.) Shorter described what he was looking for as "buried resonance", since these were too narrow and too small to show up on a swept-sinewave display - and they were real, measurable, and clearly audible, not just an academic hand-waving hypothesis. These techniques formed the basis of more than a decade's worth of BBC monitors.
Shorter's method used a slowly swept sinewave generator with an external gate that counted the sinewaves and chopped them on and off with a user-selectable 4, 8, and 16 on-and-off duty cycle, with the provision of extending the "off" interval as long as desired. The scope is then triggered by the on-off signal pulses (a secondary output of the switcher). In practice, the apparatus requires a time-of-flight delay for the switching pulse for the scope trigger. The area of the scope display that is of interest, of course, is the "off" interval. As the sinewave generator is slowly swept across the spectrum, certain frequencies show high-Q, quite narrow resonances that can be only a few tens of Hz wide. With a sparse musical spectra, these are rarely stimulated, but with a dense spectra (like a chorus or an orchestra), they can be quite audible.
I used a system like this at Audionics in 1976 minus the time-of-flight delay, so it was only good for looking at the current going through the voice coil. Even so, I was able to clearly see not one, not two, but three narrow resonances in the KEF B110 in the vicinity of 3.2~3.5 kHz - as I recall, they were around 3.2, 3.3, and 3.4 kHz, and were about 30~50 Hz wide each. These were dustcap resonances, of course, but simply removing the dustcap didn't get rid of them - new ones appeared elsewhere. (You always gotta measure, not just assume!)
Similarly, the KEF B139 bass driver had a humongous resonance at 1.5 khz from the expanded foam diaphragm - and I was aware of no loudspeaker that used a notch filter to remove it. I found that with my first loudspeaker, the Audionics TLM-200, that it required a 3rd-order lowpass filter at 200 Hz to remove the midrange coloration from audibility - 2nd-order was not enough, at least with pink-noise test stimulus. Yes, the filter was expensive, but flip the switch, the coloration would obviously come and go.
Note that all of the measurements I'm speaking of here - CSD and Shorter's chopped-sinewave method - look at what happens when the electrical stimulation is cut off.
No relationship to BL product, cone mass, and rise time at all - these techniques look at the behavior of the mechanical system, which is what we care about most, since the correction techniques are so limited in scope.
Yes, you can use notch filters, but I can say from experience they don't completely remove the original coloration (despite what measurements may say), and there is a subtle residue of what I call "grayness" to the resulting sound. (Instrumental tone colors are subtly diminished after aggressive notch-filter equalization.) The best approach to remove these gremlins is to modify the mechanical system (or do the right thing and select another driver).
What truly surprises me is that design methodologies that became standard practice across the BBC (a world-famous organization that publishes in English!) during the late Sixties never found their way into the US high-efficiency manufacturers - Altec, JBL, Electro-Voice, and University. This seems like a country determinedly clinging to black-and-white TV just because their engineers couldn't do color, or more to the point, the surprising absence of plastic-based AC-biased magnetic-tape recording technology in the English-speaking world before WWII. (Prewar magnetic recording in the USA and UK was limited to steel wire and DC bias, with very low-fi results. Low-noise, hifi recording was limited to first-generation acetate masters.)
My web page is a once-over-lightly on what you can expect to see when the impulse response is transformed to a CSD and FR plot. My favorite odd duck is the Eton CSD, shown below. I've never seen this behavior before - the obvious resonance (that long mountain range) is actually frequency-modulated! Remarkable! That shouldn't even be possible, if you stick to lumped-element theory. The real world is always a lot more complicated than our simplistic models - and the deeper you look, with more powerful measurement techniques, the more you find.
The original CSD was D.E.L. Shorter's chopped-sinewave method described in Wireless World in the early Sixties. I wish I had copies of these articles - they were as significant as Theile/Small theory, but were not appreciated in the USA when they came out. (That transatlantic accent, y'know, along with a good dose of NIH in the American speaker industry.) Shorter described what he was looking for as "buried resonance", since these were too narrow and too small to show up on a swept-sinewave display - and they were real, measurable, and clearly audible, not just an academic hand-waving hypothesis. These techniques formed the basis of more than a decade's worth of BBC monitors.
Shorter's method used a slowly swept sinewave generator with an external gate that counted the sinewaves and chopped them on and off with a user-selectable 4, 8, and 16 on-and-off duty cycle, with the provision of extending the "off" interval as long as desired. The scope is then triggered by the on-off signal pulses (a secondary output of the switcher). In practice, the apparatus requires a time-of-flight delay for the switching pulse for the scope trigger. The area of the scope display that is of interest, of course, is the "off" interval. As the sinewave generator is slowly swept across the spectrum, certain frequencies show high-Q, quite narrow resonances that can be only a few tens of Hz wide. With a sparse musical spectra, these are rarely stimulated, but with a dense spectra (like a chorus or an orchestra), they can be quite audible.
I used a system like this at Audionics in 1976 minus the time-of-flight delay, so it was only good for looking at the current going through the voice coil. Even so, I was able to clearly see not one, not two, but three narrow resonances in the KEF B110 in the vicinity of 3.2~3.5 kHz - as I recall, they were around 3.2, 3.3, and 3.4 kHz, and were about 30~50 Hz wide each. These were dustcap resonances, of course, but simply removing the dustcap didn't get rid of them - new ones appeared elsewhere. (You always gotta measure, not just assume!)
Similarly, the KEF B139 bass driver had a humongous resonance at 1.5 khz from the expanded foam diaphragm - and I was aware of no loudspeaker that used a notch filter to remove it. I found that with my first loudspeaker, the Audionics TLM-200, that it required a 3rd-order lowpass filter at 200 Hz to remove the midrange coloration from audibility - 2nd-order was not enough, at least with pink-noise test stimulus. Yes, the filter was expensive, but flip the switch, the coloration would obviously come and go.
Note that all of the measurements I'm speaking of here - CSD and Shorter's chopped-sinewave method - look at what happens when the electrical stimulation is cut off.
No relationship to BL product, cone mass, and rise time at all - these techniques look at the behavior of the mechanical system, which is what we care about most, since the correction techniques are so limited in scope.
Yes, you can use notch filters, but I can say from experience they don't completely remove the original coloration (despite what measurements may say), and there is a subtle residue of what I call "grayness" to the resulting sound. (Instrumental tone colors are subtly diminished after aggressive notch-filter equalization.) The best approach to remove these gremlins is to modify the mechanical system (or do the right thing and select another driver).
What truly surprises me is that design methodologies that became standard practice across the BBC (a world-famous organization that publishes in English!) during the late Sixties never found their way into the US high-efficiency manufacturers - Altec, JBL, Electro-Voice, and University. This seems like a country determinedly clinging to black-and-white TV just because their engineers couldn't do color, or more to the point, the surprising absence of plastic-based AC-biased magnetic-tape recording technology in the English-speaking world before WWII. (Prewar magnetic recording in the USA and UK was limited to steel wire and DC bias, with very low-fi results. Low-noise, hifi recording was limited to first-generation acetate masters.)
Attachments
measuring 'colorations"
So you seem confident that CSD's allow you to correlate measured imperfections in the response of drivers to audible artifacts with a high level of confidence? Are there circumstances where the perceived quality of a driver may not correlate with it's measured response, and how would you resolve that conflict? And what sort of limits would you impose on a CSD wrt low level disturbances and their audibility?
Lots of questions, but I'm curious as to how one best characterizes drivers beyond your own hearing so that you can have confidence you're not fooling yourself..
thnx...
John L.
So you seem confident that CSD's allow you to correlate measured imperfections in the response of drivers to audible artifacts with a high level of confidence? Are there circumstances where the perceived quality of a driver may not correlate with it's measured response, and how would you resolve that conflict? And what sort of limits would you impose on a CSD wrt low level disturbances and their audibility?
Lots of questions, but I'm curious as to how one best characterizes drivers beyond your own hearing so that you can have confidence you're not fooling yourself..
thnx...
John L.
Re: measuring 'colorations"
Well, let's put it this way. I didn't come up with this protocol; the BBC did some forty-plus years ago. I've been to Britain, visited the BBC Research Labs, and heard their most advanced recording and playback systems. To this day that was some of the best sound I've ever heard. So I'm partial to the BBC/Spendor/Quad philosophy, both esthetically and sonically. Other people prefer other schools of design.
I don't think there's anything magical or mystical about speaker design; their "personality", if you will, simply reflects the priorities of that school of the design, the skill and tastes of the designer, and the attention to detail. That's really all there is to it.
Different schools of design sound different; this isn't surprising considering how bad loudspeakers are in objective terms, and designers are forced to trade one set of virtues against others. It's the same as cars: a Porsche 911, a Hummer H1, a Ford F250 with dual rear tires, a Subaru Outback, and a Prius all do different things, and no, you can't combine them into a "supercar."
The claim that any speaker can "do it all" is simply absurd - it usually reflects a young, just-starting-out designer who thinks they are smarter than anyone else in the world. I lost that illusion about thirty years ago - I'm just another worker in the vineyard, tending the grapes, and hoping for a good harvest.
As for sonic correlation, well, you win some, you lose some. I have pretty high confidence in time and CSD data, and I have plenty of company in the industry. But taking just one example, fairly audible diffraction effects don't show up that much in the CSD, yet if you walk around the speaker while playing pink-noise, you can hear the little secondary sources coming from the cabinet edges. And physically radiusing the edges and covering them with felt may measure the same (on the CSD), but they sound quite different in spatial-rendition terms.
Crossover caps are the most annoying of all; the measured differences in DA and DF are really really small, yet the subjective effect can be as large as physically replacing drivers. I have no explanation for that at all. I think JBL's idea of applying 9V DC battery bias to a pair of caps is a good one, though - anything to reduce this annoying coloration.
P.S. Sorry about the lack of info on the new project - I'm doing a lot of off-line research and co-ordination with other folks on compression drivers and horns. As for the comment a few days ago - "why no patents on plastic-film-diaphragm compression drivers?" Well, good question, but from what I can see, Altec and JBL didn't do a lot of research on these - this seems to be a European thing. So, for all of you eager beavers out there who can read more languages than I can, if you can dig up the original B&C, Beyma, and BMS patents, why not post them here?
auplater said:
Well, let's put it this way. I didn't come up with this protocol; the BBC did some forty-plus years ago. I've been to Britain, visited the BBC Research Labs, and heard their most advanced recording and playback systems. To this day that was some of the best sound I've ever heard. So I'm partial to the BBC/Spendor/Quad philosophy, both esthetically and sonically. Other people prefer other schools of design.
I don't think there's anything magical or mystical about speaker design; their "personality", if you will, simply reflects the priorities of that school of the design, the skill and tastes of the designer, and the attention to detail. That's really all there is to it.
Different schools of design sound different; this isn't surprising considering how bad loudspeakers are in objective terms, and designers are forced to trade one set of virtues against others. It's the same as cars: a Porsche 911, a Hummer H1, a Ford F250 with dual rear tires, a Subaru Outback, and a Prius all do different things, and no, you can't combine them into a "supercar."
The claim that any speaker can "do it all" is simply absurd - it usually reflects a young, just-starting-out designer who thinks they are smarter than anyone else in the world. I lost that illusion about thirty years ago - I'm just another worker in the vineyard, tending the grapes, and hoping for a good harvest.
As for sonic correlation, well, you win some, you lose some. I have pretty high confidence in time and CSD data, and I have plenty of company in the industry. But taking just one example, fairly audible diffraction effects don't show up that much in the CSD, yet if you walk around the speaker while playing pink-noise, you can hear the little secondary sources coming from the cabinet edges. And physically radiusing the edges and covering them with felt may measure the same (on the CSD), but they sound quite different in spatial-rendition terms.
Crossover caps are the most annoying of all; the measured differences in DA and DF are really really small, yet the subjective effect can be as large as physically replacing drivers. I have no explanation for that at all. I think JBL's idea of applying 9V DC battery bias to a pair of caps is a good one, though - anything to reduce this annoying coloration.
P.S. Sorry about the lack of info on the new project - I'm doing a lot of off-line research and co-ordination with other folks on compression drivers and horns. As for the comment a few days ago - "why no patents on plastic-film-diaphragm compression drivers?" Well, good question, but from what I can see, Altec and JBL didn't do a lot of research on these - this seems to be a European thing. So, for all of you eager beavers out there who can read more languages than I can, if you can dig up the original B&C, Beyma, and BMS patents, why not post them here?
This 15"er is snuck away in the Eminence range.
http://www.eminence.com/pdf/kappalite-3015lf.pdf
Reasonable price, so two in parallel might be useful for OB use ?
Cheers ........ Graham.
http://www.eminence.com/pdf/kappalite-3015lf.pdf
Reasonable price, so two in parallel might be useful for OB use ?
Cheers ........ Graham.
Re: Re: measuring 'colorations"
Personally I believe the "sound" of caps may be due to unrecognized dimensions to the loss tangent and/or dielectric constant. Since the permittivity and the loss tangent assign numerical values to the charge capacity and "mobility" of the dipoles within the dielectric respectively, there may be anisotropic motion not clearly represented by these two values; i.e., "steric" hindrance in rotation or relaxation about one or more axes perhaps, depending on the material. I dunno.. if I were younger I might try figuring this out experimentally, I know the rf engineers i used to work with were all pretty touchy about parasitic capacitance...
thanx for the info.
John L.
Lynn Olson said:
Personally I believe the "sound" of caps may be due to unrecognized dimensions to the loss tangent and/or dielectric constant. Since the permittivity and the loss tangent assign numerical values to the charge capacity and "mobility" of the dipoles within the dielectric respectively, there may be anisotropic motion not clearly represented by these two values; i.e., "steric" hindrance in rotation or relaxation about one or more axes perhaps, depending on the material. I dunno.. if I were younger I might try figuring this out experimentally, I know the rf engineers i used to work with were all pretty touchy about parasitic capacitance...
thanx for the info.
John L.
Re: Re: measuring 'colorations"
PS: JBL now buy in BMS CD's for some of their speakers.
This is likely a response to my post. I wasn't asking you to do research, merely asking for any opinions and or insights into these diaphram materials you may have picked up over time as I find your comments often insightful and worthy of further investigation on my own. The post was merely an observation that all your comments related to metallic or phenolic diaphrams.Lynn Olson said:
PS: JBL now buy in BMS CD's for some of their speakers.
My comments on diaphragm materials aren't especially brilliant or profound. If we could take these compression drivers, unscrew the back cap, knock out the phase-plug with a mallet (Alexander's suggestion), and turn them around and listen to the naked diaphragm, what do we hear?
Well, even "small-format" domes are pretty big at 1.75", and traditional large-format diaphragms look downright huge at 2.88" and 4". Compared to direct-radiators, these are big domes. The suspensions vary from tangential, to diamond, to the same kind of thing we see in dome tweeters and midranges, a corrugated roll of compliant plastic or treated fabric.
Compression drivers don't get any exemptions from the laws of physics. The bigger the dome, the lower the breakup frequencies, and this is directly correlated to size. Damping techniques merely damp resonances, but don't do all that much to change the frequencies where breakup starts.
So if an audiophile metal dome that's 1-inch across starts to go into uncontrolled breakup in the 22~26 kHz region, we shouldn't expect wonders from a diaphragm made of the same materials, with similar suspension, that is three to four times bigger, and has nine to sixteen times the area. That's why I am so skeptical of large-format drivers operating up to 20 kHz - what direct-radiator of that size would be free from breakup? I can't think of any.
Well, even "small-format" domes are pretty big at 1.75", and traditional large-format diaphragms look downright huge at 2.88" and 4". Compared to direct-radiators, these are big domes. The suspensions vary from tangential, to diamond, to the same kind of thing we see in dome tweeters and midranges, a corrugated roll of compliant plastic or treated fabric.
Compression drivers don't get any exemptions from the laws of physics. The bigger the dome, the lower the breakup frequencies, and this is directly correlated to size. Damping techniques merely damp resonances, but don't do all that much to change the frequencies where breakup starts.
So if an audiophile metal dome that's 1-inch across starts to go into uncontrolled breakup in the 22~26 kHz region, we shouldn't expect wonders from a diaphragm made of the same materials, with similar suspension, that is three to four times bigger, and has nine to sixteen times the area. That's why I am so skeptical of large-format drivers operating up to 20 kHz - what direct-radiator of that size would be free from breakup? I can't think of any.
Lynn Olson said:
Exactly right.
A 1" aluminum dome breaks up at roughly 25 kHz. A 2" aluminum dome breaks up at roughly half that.
So when you are looking at a 4" large format compression driver, the diaphragm is breaking up in the 5 - 6 kHz range. Past that and all you have are a myriad of resonances.
The smallest of the "small format" compression drivers have 1.75" domes. So if they are of aluminum or titanium, they will go up to around 15 kHz, best case. The only way around this is to use beryllium. That is why the TAD 2001 (designed by Bart Locanthi, one of the true audio geniuses) is so well regarded among certain cognoscenti -- it will go to around 22 kHz before breakup.
And you can use it down to around 1 kHz. So this is a good solution, as long as you don't care about money. The driver is over $1000 each, and a nice wooden horn is around $300 a pair just for starters.