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Post by mgbpuff on Mar 11, 2024 10:01:28 GMT -5
Lots of complicated statements here. But we never want amplifier clipping, period. We never want speaker bottoming out, period. We never want mismatched amp / speaker pairings (the amplifier power and the speaker compliance must compliment each other). The first watt statement is valid only because most of our comfortable listening levels of normal source material require very little sound power delivered to our ear drums (fact). Higher voltage enables more control over reactive loads than lower voltage (fact - basic engineering science truth)! A given speaker does not consume more power to deliver a given sound level regardless of the amplifier rating provided that the amp is not clipping. Lets not talk about pro equipment as they are generally designed to deliver a lot of power to extremely large spaces and accuracy is not as treasured in such environments.
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KeithL
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Post by KeithL on Mar 11, 2024 10:16:17 GMT -5
I don't specifically disagree with any of this... However there is one little thing that I would add... which is that, as with most things, there are trade-offs when it comes to both loudspeaker and individual driver design. And, with loudspeakers, those tradeoffs almost include a lot of things, and price may or may not even be one of them. It's not as if you can "just pay a little more and get a speaker that sounds equally good but is much more efficient". Just as you cannot "pay a bit more and get a Rolls-Royce that drives the same but gets 500 miles-per-gallon". Unfortunately, when it comes to the design of speaker drivers, accuracy and low distortion tend to come at the cost of efficiency and size. It used to be a sort of truism that more powerful magnets would be both more expensive and give you higher efficiency. However, while this is still true to a degree, even incredibly powerful modern magnets don't cost all that much. I really hate making generalizations but... to be quite blunt... There are very few "super high efficiency speakers" available these days... And MOST "audiophile quality" speakers tend to be of low-to-medium efficiency... And, yes, part of the reason is that "amplifier power is very cheap" compared to what it cost "in the good old days"... You will also notice that most pro drivers, rather than being "super-efficient", are simply rated to be both "relatively efficient", and also able to handle massive amounts of power. But, again, note that many of them are not considered to sound good enough to be used in home audiophile speakers. (I would also point out that, anecdotally, the sound systems at the last several concerts I attended, in large venues, did NOT sound as good as what I have in my living room.) Using a more powerful amp than what one might deem absolutely necessary, may pay dividends in dynamic range (if the source supports wide dynamics) which can add to the sense of realism. Of course, the speakers are a big part of this equation also. People seem to be very tolerant of compressed music whether it is compressed at mastering or compressed by modest amplification and speaker performance. Audiophiles want music to get as realistic sounding as possible. There's no one answer for all combinations of environment, equipment, and desires. A more efficient speaker may pay dividends in dynamic range. Unclipped modest power or high power amplifiers with similar circuitry, driven to the same level, produce the same dynamic range, into the same load. A low power amplifier with the 'right stuff' can sound better than a high power amplifier - power supply, first watt quality, reasonable damping difference, output current capability... High power with(because of) higher voltage rails means nothing by itself. An amplifier is a liner device. The higher power amplifier can produce more unclipped volume, but, as more power is consumed by the speaker, the speaker becomes more of a compressor. ALL speakers begin to compress at relatively low power inputs. High power input to any real speaker, and more excursion, creates more compression. The first key to realistic dynamic range is speaker efficiency. Check out pro driver sites. Compression specs for various input power are often listed. You won’t see this spec for any consumer speaker. You can also see this graphically in some of the Klippel data for drivers. The most dynamic speaker is one that consumes no power when playing. They don't exist. Some get closer than others. The most dynamic amplifier is one that remains in 'stasis' regardless the load. Hmmm... where have we seen that term before.
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Post by PaulBe on Mar 11, 2024 11:23:56 GMT -5
I don't specifically disagree with any of this... However there is one little thing that I would add... which is that, as with most things, there are trade-offs when it comes to both loudspeaker and individual driver design. And, with loudspeakers, those tradeoffs almost include a lot of things, and price may or may not even be one of them. It's not as if you can "just pay a little more and get a speaker that sounds equally good but is much more efficient". Just as you cannot "pay a bit more and get a Rolls-Royce that drives the same but gets 500 miles-per-gallon". Unfortunately, when it comes to the design of speaker drivers, accuracy and low distortion tend to come at the cost of efficiency and size. It used to be a sort of truism that more powerful magnets would be both more expensive and give you higher efficiency. However, while this is still true to a degree, even incredibly powerful modern magnets don't cost all that much. I really hate making generalizations but... to be quite blunt... There are very few "super high efficiency speakers" available these days... And MOST "audiophile quality" speakers tend to be of low-to-medium efficiency... And, yes, part of the reason is that "amplifier power is very cheap" compared to what it cost "in the good old days"... You will also notice that most pro drivers, rather than being "super-efficient", are simply rated to be both "relatively efficient", and also able to handle massive amounts of power. But, again, note that many of them are not considered to sound good enough to be used in home audiophile speakers. (I would also point out that, anecdotally, the sound systems at the last several concerts I attended, in large venues, did NOT sound as good as what I have in my living room.) A more efficient speaker may pay dividends in dynamic range. Unclipped modest power or high power amplifiers with similar circuitry, driven to the same level, produce the same dynamic range, into the same load. A low power amplifier with the 'right stuff' can sound better than a high power amplifier - power supply, first watt quality, reasonable damping difference, output current capability... High power with(because of) higher voltage rails means nothing by itself. An amplifier is a liner device. The higher power amplifier can produce more unclipped volume, but, as more power is consumed by the speaker, the speaker becomes more of a compressor. ALL speakers begin to compress at relatively low power inputs. High power input to any real speaker, and more excursion, creates more compression. The first key to realistic dynamic range is speaker efficiency. Check out pro driver sites. Compression specs for various input power are often listed. You won’t see this spec for any consumer speaker. You can also see this graphically in some of the Klippel data for drivers. The most dynamic speaker is one that consumes no power when playing. They don't exist. Some get closer than others. The most dynamic amplifier is one that remains in 'stasis' regardless the load. Hmmm... where have we seen that term before. Nothing I said was mean't to be idealistic. There are trade-offs in everything. Plenty of micro issues to argue about. Let's keep it simple... KISS...? Loudspeakers are compressors. ALL of them. Big and small amplifiers are linear. Big amplifiers Don't produce better dynamics. Speakers driven with large amounts of power compress more than when driven with less amounts of power. Efficient speakers driven at equal volumes to less efficient speakers compress less. Part of the reason MOST audiophile speakers are low-to-medium efficiency is because power is soaked up in the crossovers to smooth frequency response, which is why an expensive Revel Salon can be so flat but has poor efficiency AND good dynamics. Pro drivers sound very good. They have to sound good or they wouldn't be bought. Livelihoods depend on them. Perhaps you could define your difference between 'super' and 'relative' efficiency. Your terms don't relate to much. However, I've never heard a live venue where the system sounded as good as what I use at home, nor do I use xxxxKW of power to drive my system, nor is my home a 2 acre concert hall or stadium. But, when some of these pro drivers are used in a home system... well... there's where you separate the men from the consumer boys. The pro drivers are still loafing when the consumer driver is suffering from emphysema. I care more about real performance than what is considered in audiophile circles. Audiophoolery is as much a 'driver of the month' club as there is a 'boutique opamp of the month' club. More KISS... most home speakers are in the 87-90dB/1 watt range because of physics/economics. Lot of competition for the same space and dollar. I get it. They aren't 87-90dB efficient because this range is inherently better for some reason. It's a trade off... with calculated variables. Why can't we just agree that ALL speakers are compressors, and amplifiers are linear? That was my factual point. FTR - I use 15" woofers that are used in pro monitoring applications and in custom high end home systems; AETD15S, Apollo version, 95dB/2.83v, 28mm linear p-p, 1KW power handling ability. I suppose they would be termed 'relative' high efficiency. I'm glad I bought them before the 40% price increase that occurred last Fall.
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KeithL
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Post by KeithL on Mar 11, 2024 11:25:44 GMT -5
Now you're getting into a more useful and practical area. In order of "practical significance"... the first/best part of the passive crossover to get rid of is the part on the woofer. The woofer is typically going to use the most power on average. However, and far more importantly, the woofer is usually the heaviest driver, typically being a large cone, and has a powerful motor structure. Because of this the woofer is also the driver that is most likely to both require and benefit from "being tightly controlled by the amplifier" (damping). Since the passive components in the crossover actually reduce the damping slightly - having the amplifier connected directly to the woofer is a benefit in this regard. Next up would be the midrange... which may also benefit significantly from being directly connected to the amplifier. While the tweeter may get some benefit from being directly connected to the amplifier it would receive the least. Therefore there is little justification in using a separate amplifier for the tweeter... (Except in very large and powerful "venue installations".) For this reason, with bi-amped three-way speakers, it is common to connect one amplifier directly to the woofer... And connect the other directly to the midrange... and also to the tweeter through a crossover... Elliot Sound Products article conceringing ACTIVE biamp......That is? Where the frequency is divided prior to the amps and is distributed than. Speaker level crossover is mostly deleted....except maybe the mid-tweet....... Passive has none of the advantages of 'Active'...where each amp gets the full 20->20.......and it is manipulated at the speaker level. sound-au.com/bi-amp.htmMorgan? I think you are in range of Tampa. Head over to YBOR City for dinner at the Colombia. I hope it is still 'good'. But pack some heat. Crime Rate in about 300% the national average. And you can't get a locally wrapped Cigar, any more. Early 70's?? I'd go to the Perfecto Garcia plant and pick up a box of Cigarellos.....freshly rolled....
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Post by mgbpuff on Mar 11, 2024 11:42:41 GMT -5
The real failure of all present audio systems today is the lack of feedback. I am not speaking of interim circuit feedback which, being easy to implement, is widely used. What I mean is final output feedback, the audio wave itself. This could be some form of microphonic feedback or perhaps some form of movement feedback. Close the loop on it all and perhaps we will finally have the real solution to high resolution audio.
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Post by marcl on Mar 11, 2024 11:43:41 GMT -5
Now you're getting into a more useful and practical area. In order of "practical significance"... the first/best part of the passive crossover to get rid of is the part on the woofer. The woofer is typically going to use the most power on average. However, and far more importantly, the woofer is usually the heaviest driver, typically being a large cone, and has a powerful motor structure. Because of this the woofer is also the driver that is most likely to both require and benefit from "being tightly controlled by the amplifier" (damping). Since the passive components in the crossover actually reduce the damping slightly - having the amplifier connected directly to the woofer is a benefit in this regard. Next up would be the midrange... which may also benefit significantly from being directly connected to the amplifier. While the tweeter may get some benefit from being directly connected to the amplifier it would receive the least. Therefore there is little justification in using a separate amplifier for the tweeter... (Except in very large and powerful "venue installations".) For this reason, with bi-amped three-way speakers, it is common to connect one amplifier directly to the woofer... And connect the other directly to the midrange... and also to the tweeter through a crossover... Elliot Sound Products article conceringing ACTIVE biamp......That is? Where the frequency is divided prior to the amps and is distributed than. Speaker level crossover is mostly deleted....except maybe the mid-tweet....... Passive has none of the advantages of 'Active'...where each amp gets the full 20->20.......and it is manipulated at the speaker level. sound-au.com/bi-amp.htmMorgan? I think you are in range of Tampa. Head over to YBOR City for dinner at the Colombia. I hope it is still 'good'. But pack some heat. Crime Rate in about 300% the national average. And you can't get a locally wrapped Cigar, any more. Early 70's?? I'd go to the Perfecto Garcia plant and pick up a box of Cigarellos.....freshly rolled.... In the Magnepan world (and I suspect other speakers too) there is constant debate about "upgrading" the capacitors and inductors in the passive crossover. Recently Magnepan introduced their "X" models (also available as a very expensive upgrade to current models). I guess "X" is this year's speaker upgrade moniker! The Magnepan X models include a number of mechanical and electrical changes. These include hand-wound air core inductors and hand measured and matched upgraded capacitors in the passive crossovers. It's also really interesting that Magnepan chose to reveal all of the aspects of the X models/upgrade as opposed to the past (3.7 and 1.7 "i" and LRS "+") where they did not reveal what they did. So in the context of your comments and the value of biamping and eliminating the passive crossover .... what are your thoughts on passive crossover "upgrades" vs active crossovers?
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Post by PaulBe on Mar 11, 2024 11:50:18 GMT -5
Lots of complicated statements here. But we never want amplifier clipping, period. We never want speaker bottoming out, period. We never want mismatched amp / speaker pairings (the amplifier power and the speaker compliance must compliment each other). The first watt statement is valid only because most of our comfortable listening levels of normal source material require very little sound power delivered to our ear drums (fact). Higher voltage enables more control over reactive loads than lower voltage (fact - basic engineering science truth)! A given speaker does not consume more power to deliver a given sound level regardless of the amplifier rating provided that the amp is not clipping. Lets not talk about pro equipment as they are generally designed to deliver a lot of power to extremely large spaces and accuracy is not as treasured in such environments. ALL speakers are compressors, and amplifiers are linear. That was my Factual point and my Only point. There is NO relationship between amplifier power and the speaker compliance. The first watt statement is valid for ALL listening situations. More voltage has NOTHING to do with handling reactive loads. What is needed in an amplifier is a good power supply, a large SOA in the output stage, and a stable feedback loop. Facts... I said "Unclipped modest power or high power amplifiers with similar circuitry, driven to the same level, produce the same dynamic range, into the same load.". You said "A given speaker does not consume more power to deliver a given sound level regardless of the amplifier rating provided that the amp is not clipping.". Our Respective Statements Are Compatible. Problem? I said check out pro sites for driver compression specs. You won't find these specs at consumer sites. All drivers compress. You will see examples of compression in good pro drivers. Consumer drivers have much worse compression. No consumer manufacturer will give you compression specs. Accuracy is 'treasured' much more in pro environments than is generally 'treasured' in consumer environments. The 'pro drivers are bad' argument is a myth. I offered this to help you understand that high power amplifiers Don't produce better dynamics. In FACT, they can create more compression when used at High Power. ALL speakers are compressors, and amplifiers are linear. That was my Factual point and my only point.
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KeithL
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Post by KeithL on Mar 11, 2024 11:50:38 GMT -5
I'm going to just add a little bit of clarification here... In modern home high-fidelity systems we absolutely prefer to avoid clipping if possible. And, with the current availability of quite powerful amps at reasonable prices, this is not at all difficult to achieve. (And, if the occasional drumbeat clips for a split second, most modern amplifiers will clip cleanly and it won't be audible.) However, in extremely large professional installations, like in large concert halls... And especially in the sort often used for loud rock concerts (and often played at truly ear-threatening levels)... Clipping is both normal and expected... (although, again, less so with modern limiters and compressors). In those situations, a lot of power is needed, and "getting the most out of every watt" is a serious consideration. And, in those situations, it is considered to be a major benefit to prevent clipping in one set of drivers from audibly affecting other drivers. Specifically, being able to run the woofers into hard clipping, without seriously "muddying up the midrange". (Also note that, while clipping home speakers is also very bad for the speakers, "club speakers" and "instrument speakers" are usually designed to tolerate it.) Also note that this situation was "more true" for home systems as well "in the old days"... Back when powerful amplifiers were expensive, and most amps were actually relatively low in power, and especially with old tube equipment, which often "clips gracefully"... To have one amplifier that could continue to deliver clean power to the midrange and tweeter "after the woofer started to clip a little bit"... (Back then there was a big jump in the price of amplifiers above the 50-100 watt/channel range....) I would correct you on one thing... Higher voltage would enable you to drive more power into a highly reactive load because of its proportionally higher impedance... So having higher rails may enable you to drive a highly reactive load better because it requires more voltage for a given amount of power. Also, at least in theory, it may enable the output devices to operate more linearly "because they are operating over a smaller portion of their operating range". However, in order to be able to control a load effectively, it it is the ability to sink or cancel out the current that results from their back-EMF that matters. (That's "damping factor" - which is related to "the source impedance the load sees looking back into the amplifier".) Lots of complicated statements here. But we never want amplifier clipping, period. We never want speaker bottoming out, period. We never want mismatched amp / speaker pairings (the amplifier power and the speaker compliance must compliment each other). The first watt statement is valid only because most of our comfortable listening levels of normal source material require very little sound power delivered to our ear drums (fact). Higher voltage enables more control over reactive loads than lower voltage (fact - basic engineering science truth)! A speaker does not consume more power to deliver a given sound level regardless of the amplifier rating provided that the amp is not clipping. Lets not talk about pro equipment as they are generally designed to deliver a lot of power to extremely large spaces and accuracy is not as treasured in such environments.
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KeithL
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Post by KeithL on Mar 11, 2024 12:00:29 GMT -5
The "catch" is in WHERE you sample your feedback. Velodyne made some pretty impressive subwoofers in the past - that had actual sensors attached to the cone. However this isn't really practical on smaller cones or the various different types of tweeters. Bear in mind that this sort of correction is NOT exactly EQ. You're talking about correcting the actual shape of the waveform... while the driver is in motion. I'll also let you in on a little secret (not really)... We tend to model things like cones and domes as if they were "perfect pistons" or some such... And we assume that, when we apply a signal to the voice coil, the voice coil follows it exactly, and the surface follows the voice coil exactly as well... Unfortunately the reality is very far from this... (Well, it can be pretty close, for a big subwoofer, with a really stiff cone, at relatively low frequencies.) So, to put it bluntly, having an exact measurement of where one spot on that tweeter is won't tell you much about a spot even a few millimeters from that spot. (And remember that, unless you can be within a few percent of dead accurate, your correction is going to cause more distortion than it corrects.) (So, even if you could measure two dozen spots on that tweeter, unless you could correct each one separately, you would cause more harm than good.) In theory this could be done with some sort of planar surface... With both drive and sensors distributed across its surface... And such things are being done... but only at a very small scale... and over very limited frequencies. (Like "phased arrays" for ultrasonic transducers.) (You can also make a big array, out of hundreds of tweeters, and treat each entire tweeter as "one element in the array".) But I don't see a speaker that works this way showing up any time soon. The real failure of all present audio systems today is the lack of feedback. I am not speaking of interim circuit feedback which, being easy to implement, is widely used. What I mean is final output feedback, the audio wave itself. This could be some form of microphonic feedback or perhaps some form of movement feedback. Close the loop on it all and perhaps we will finally have the real solution to high resolution audio.
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KeithL
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Post by KeithL on Mar 11, 2024 12:25:42 GMT -5
Now, to me, THAT is a very interesting question... Compared to a cone driver... the "motor assembly" on Magneplanars is VERY loosely coupled. Therefore I would expect far less interaction between the amplifier and the speaker. I wouldn't expect things like the damping factor of the amp to matter much... and Magnepans are a relatively non-reactive load. (I don't know if that's true for the range covered by the tweeter in the models that have a true ribbon tweeter.) So I wouldn't expect any improvement due to "tighter coupling" or "better damping" if the crossovers were eliminated. So that, in turn, makes it more important what "direct effect" the crossovers are having on the sound... And whether "upgrades" were really "improvements" or not. For example... take replacing a metal-core inductor with an air-core inductor. NORMALLY... - we would look at the resistance... which might be lower with the metal core inductor... - but we would also look at potential nonlinearities... which might be better with the air core inductor BUT... - because the Magnepan is less sensitive to damping it would see LESS BENEFIT from lowering that resistance - but it might still benefit from better linearity With capacitors I would say that it would be more of the proverbial crap shoot... Some capacitors sound noticeably bad... but many do not... So it would depend on the specifics of the ones chosen (and the ones currently there). However, in general, I'm inclined to thing that eliminating parts is more likely to eliminate problems than cause them. As for their "upgrades"... I have to give that a big "it depends"... For example "hand measured inductors" sounds cool... But it's only an improvement if they're more accurate than before (and especially if they had a known issue with off-spec parts)... So it could be that before they were hiding "fixing problems" or simple production changes as "unspecified upgrades"... (Maybe they just bought good poly caps from whoever was cheaper that week and it really doesn't make any difference.) Or it could be that they've decided that spelling out the details plays better in the marketing literature than simply saying "new and improved"... Or it could be that this is a response to too many people becoming uneasy about "which version they have" and "why theirs are different"... To be honest... any of these makes sense to me... so your guess is as good as mine. A LOT also depends on the rest of your system.... For example, since YOU are already using DSPs and the like, the crossovers are just adding a level of complexity... And eliminating them would tend to simplify the overall situation... Which I suspect would be a plus... However, for someone who is NOT using DSP filters, and room correction... Tweaking the crossover could make a big difference... or could simply have become necessary because of other production changes... (But, to be blunt, I think the way Maggies sound is pretty subjective to begin with, so I would want to hear for myself before making any assumptions.) To me there are a lot more things likely to affect the way they sound, ranging from how thick the plastic film is, to how tightly you stretch it. That's why I would expect a lot of variability from model to model, and batch to batch, and possibly even between production runs... Which, to me, means that the way they sound is also likely to be affected more by things like crossover components than other speakers would be. ................................ In the Magnepan world (and I suspect other speakers too) there is constant debate about "upgrading" the capacitors and inductors in the passive crossover. Recently Magnepan introduced their "X" models (also available as a very expensive upgrade to current models). I guess "X" is this year's speaker upgrade moniker! The Magnepan X models include a number of mechanical and electrical changes. These include hand-wound air core inductors and hand measured and matched upgraded capacitors in the passive crossovers. It's also really interesting that Magnepan chose to reveal all of the aspects of the X models/upgrade as opposed to the past (3.7 and 1.7 "i" and LRS "+") where they did not reveal what they did. So in the context of your comments and the value of biamping and eliminating the passive crossover .... what are your thoughts on passive crossover "upgrades" vs active crossovers?
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Post by mgbpuff on Mar 11, 2024 13:26:56 GMT -5
I would correct you on one thing... Higher voltage would enable you to drive more power into a highly reactive load because of its proportionally higher impedance... So having higher rails may enable you to drive a highly reactive load better because it requires more voltage for a given amount of power. Also, at least in theory, it may enable the output devices to operate more linearly "because they are operating over a smaller portion of their operating range". However, in order to be able to control a load effectively, it it is the ability to sink or cancel out the current that results from their back-EMF that matters. (That's "damping factor" - which is related to "the source impedance the load sees looking back into the amplifier".) I come from an industrial motor / control background so I tend to see thing more in that perspective. A speaker is a motor to me although movement is relatively small. You can power it in one direction and allow friction or even spring force to remove the energy required to slow it down. You can power it in one direction and allow friction, spring force, and regeneration to slow it down (EMF). Or, if you need high performance, you power it both forward and in reverse by using a bidirectional power supply. Changing the amount and direction of current in a highly inductive circuit can be greatly enhanced by a voltage twice as high or more to force the current change to comply faster than normal resistive EMF discharge time. An audio push-pull amp does this to some extent, but if the voltage were raised, it would control faster and better by pure electro-magnetic force. Another difference between motor control and audio speakers is that the motor circuits are 'current' regulated and controlled vs the open loop 'voltage' control of audio amps on speakers.
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Post by marcl on Mar 11, 2024 14:07:42 GMT -5
Now, to me, THAT is a very interesting question... Compared to a cone driver... the "motor assembly" on Magneplanars is VERY loosely coupled. Therefore I would expect far less interaction between the amplifier and the speaker. I wouldn't expect things like the damping factor of the amp to matter much... and Magnepans are a relatively non-reactive load. (I don't know if that's true for the range covered by the tweeter in the models that have a true ribbon tweeter.) So I wouldn't expect any improvement due to "tighter coupling" or "better damping" if the crossovers were eliminated. So that, in turn, makes it more important what "direct effect" the crossovers are having on the sound... And whether "upgrades" were really "improvements" or not. For example... take replacing a metal-core inductor with an air-core inductor. NORMALLY... - we would look at the resistance... which might be lower with the metal core inductor... - but we would also look at potential nonlinearities... which might be better with the air core inductor BUT... - because the Magnepan is less sensitive to damping it would see LESS BENEFIT from lowering that resistance - but it might still benefit from better linearity With capacitors I would say that it would be more of the proverbial crap shoot... Some capacitors sound noticeably bad... but many do not... So it would depend on the specifics of the ones chosen (and the ones currently there). However, in general, I'm inclined to thing that eliminating parts is more likely to eliminate problems than cause them. As for their "upgrades"... I have to give that a big "it depends"... For example "hand measured inductors" sounds cool... But it's only an improvement if they're more accurate than before (and especially if they had a known issue with off-spec parts)... So it could be that before they were hiding "fixing problems" or simple production changes as "unspecified upgrades"... (Maybe they just bought good poly caps from whoever was cheaper that week and it really doesn't make any difference.) Or it could be that they've decided that spelling out the details plays better in the marketing literature than simply saying "new and improved"... Or it could be that this is a response to too many people becoming uneasy about "which version they have" and "why theirs are different"... To be honest... any of these makes sense to me... so your guess is as good as mine. A LOT also depends on the rest of your system.... For example, since YOU are already using DSPs and the like, the crossovers are just adding a level of complexity... And eliminating them would tend to simplify the overall situation... Which I suspect would be a plus... However, for someone who is NOT using DSP filters, and room correction... Tweaking the crossover could make a big difference... or could simply have become necessary because of other production changes... (But, to be blunt, I think the way Maggies sound is pretty subjective to begin with, so I would want to hear for myself before making any assumptions.) To me there are a lot more things likely to affect the way they sound, ranging from how thick the plastic film is, to how tightly you stretch it. That's why I would expect a lot of variability from model to model, and batch to batch, and possibly even between production runs... Which, to me, means that the way they sound is also likely to be affected more by things like crossover components than other speakers would be. In the Magnepan world (and I suspect other speakers too) there is constant debate about "upgrading" the capacitors and inductors in the passive crossover. Recently Magnepan introduced their "X" models (also available as a very expensive upgrade to current models). I guess "X" is this year's speaker upgrade moniker! The Magnepan X models include a number of mechanical and electrical changes. These include hand-wound air core inductors and hand measured and matched upgraded capacitors in the passive crossovers. It's also really interesting that Magnepan chose to reveal all of the aspects of the X models/upgrade as opposed to the past (3.7 and 1.7 "i" and LRS "+") where they did not reveal what they did. So in the context of your comments and the value of biamping and eliminating the passive crossover .... what are your thoughts on passive crossover "upgrades" vs active crossovers? Marketing. Magnepan's approach on the X business is to do what DIY Mod'ers have been doing for years. Plus, they have eliminated a process that allows paint overspray on the diaphragm to lower mass, and they are spending more time matching pairs by tweaking the magnet plate to diaphragm clearance. And a lot of it is - and this is the really weird one - the high end market sells speakers that don't sound any better for a lot more money, and some people WANT a speaker that costs a lot more than Maggies. I have not been able to listen to the X series and I don't know anyone who has done a side by side comparison in the same room with the same equipment. And ... you did answer the question! If I understand, the nature of Maggies means they may not benefit as much from biamping with active crossovers, so any improvements in the passive crossovers - if audible - could be more worth it ... is that right? BTW ... when I bought my Tympani 1D in '77 I listened biamped and not biamped at the dealer with the same Crown amps. I liked the sound biamped a lot better (my dad heard it too) so I bought the crossover and played them biamped for 35 years until I replaced them with the 3.7 which does not have the easy biamp jumpers.
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Post by KeithL on Mar 11, 2024 14:36:48 GMT -5
re *** Kind of yeah... although it's a bit more complicated than that. I would say that they would be unlikely to benefit as much from bi-amping itself as other speakers (because they will benefit less from having a direct connection to the amp). HOWEVER, if the current passive components are actually so bad that they are causing HARM, then eliminating them could be an improvement. (And you might get that improvement EITHER by replacing them with better components or by simply eliminating them by bi-amping.) HOWEVER, you also need to bear in mind that, for MOST PEOPLE, going from the internal passive crossovers to active bi-amping is going to entail ADDING a lot more components upstream. They're going to be adding either an analog active crossover, which is quite complex, or a DSP-based crossover, and so both an A/D and a D/A stage, along with the filters themselves. (But the situation is somewhat different for you... since you already have some of that stuff in the signal path.) For people with simpler setups there are considerations like the fact that adding active stages always adds some noise and distortion. Whereas the inductors and capacitors in a passive speaker crossover may add distortion but they won't be adding any noise. Another thing to consider is that the speakers were designed with their passive crossovers in mind. So the values they chose may include compensation for nonlinearities in the speakers themselves. So, after omitting the passive crossover, you may discover additional areas that need correction. You don't want to replace a high-resistance coil with a "better one with lower resistance"... only to have to replace the resistance you removed with a separate resistor because it "belongs there". For example it is quite common for a crossover to include a gentle notch filter, or a high-end roll-off, to compensate for a slight resonance or rising response in a driver. (You would have to look at the crossover design itself to try and figure out if that was the case or not.) I would go with "upgrading the passive crossover is probably more likely to improve things than bi-amping on those speakers - all things considered"... But I would not consider it to be a certainty... ...................................................... Marketing. Magnepan's approach on the X business is to do what DIY Mod'ers have been doing for years. Plus, they have eliminated a process that allows paint overspray on the diaphragm to lower mass, and they are spending more time matching pairs by tweaking the magnet plate to diaphragm clearance. And a lot of it is - and this is the really weird one - the high end market sells speakers that don't sound any better for a lot more money, and some people WANT a speaker that costs a lot more than Maggies. I have not been able to listen to the X series and I don't know anyone who has done a side by side comparison in the same room with the same equipment. *** And ... you did answer the question! If I understand, the nature of Maggies means they may not benefit as much from biamping with active crossovers, so any improvements in the passive crossovers - if audible - could be more worth it ... is that right? *** BTW ... when I bought my Tympani 1D in '77 I listened biamped and not biamped at the dealer with the same Crown amps. I liked the sound biamped a lot better (my dad heard it too) so I bought the crossover and played them biamped for 35 years until I replaced them with the 3.7 which does not have the easy biamp jumpers.
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KeithL
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Post by KeithL on Mar 11, 2024 15:03:16 GMT -5
True... And, in older loudspeakers, where the motors were weaker, and the coupling much looser, it was generally assumed that the speaker would provide significant mechanical damping and other similar losses. This is why the damping factor provided by the amplifier was less critical - and such speakers can even seem "overdamped" when driven by a modern amp. However, the woofers in most modern speakers are very heavy, have very powerful motors, and are very tightly coupled. They also tend to have far less loss in the spider and surround... and are often housed in smaller cabinets which provide more spring force as well. In short... they are designed to work with amplifiers that provide lots of electrical damping by sinking that back-EMF-driven current. (That's why a lot of modern speakers tend to sound "sloppy" if connected to tube amps... which provide relatively little damping.) Also bear in mind that, with speakers, the speaker itself is driven in both directions. So, yes, you are providing driving force to drive the speakers in both directions. But you are then also providing braking force in both directions... This braking force is required to brake the speaker when it attempts to overshoot in either direction... But, unlike a motor, the speaker is also "a weight and a spring force"... So, when no drive is applied, it will continue to move, to bounce back and forth at its resonant frequency, because it has stored energy that it needs to dissipate... (This is what you see on those "waterfall plots" where the speaker keeps producing sound at certain frequencies after the signal has stopped.) And the amplifier also applies braking force against that energy by sinking the back-EMF-driven current it generates. With many speakers this is so extreme that it is quite easy to hear... Just tap the woofer gently with your knuckle with nothing connected to the terminals... Then repeat this with the terminals shorted with a paper clip... With many speakers you will hear a big difference. (The paper clip applies "dynamic braking" by sinking that back-EMF-generated current.) Remember that a speaker has a permanent magnet... so it acts exactly like a generator if you "drive it in reverse". And, in this case, the momentum of the speaker mass moving provides the energy to drive it. I would correct you on one thing... Higher voltage would enable you to drive more power into a highly reactive load because of its proportionally higher impedance... So having higher rails may enable you to drive a highly reactive load better because it requires more voltage for a given amount of power. Also, at least in theory, it may enable the output devices to operate more linearly "because they are operating over a smaller portion of their operating range". However, in order to be able to control a load effectively, it it is the ability to sink or cancel out the current that results from their back-EMF that matters. (That's "damping factor" - which is related to "the source impedance the load sees looking back into the amplifier".) I come from an industrial motor / control background so I tend to see thing more in that perspective. A speaker is a motor to me although movement is relatively small. You can power it in one direction and allow friction or even spring force to remove the energy required to slow it down. You can power it in one direction and allow friction, spring force, and regeneration to slow it down (EMF). Or, if you need high performance, you power it both forward and in reverse by using a bidirectional power supply. Changing the amount and direction of current in a highly inductive circuit can be greatly enhanced by a voltage twice as high or more to force the current change to comply faster than normal resistive EMF discharge time. An audio push-pull amp does this to some extent, but if the voltage were raised, it would control faster and better by pure electro-magnetic force. Another difference between motor control and audio speakers is that the motor circuits are 'current' regulated and controlled vs the open loop 'voltage' control of audio amps on speakers.
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Post by marcl on Mar 12, 2024 7:53:02 GMT -5
True... And, in older loudspeakers, where the motors were weaker, and the coupling much looser, it was generally assumed that the speaker would provide significant mechanical damping and other similar losses. This is why the damping factor provided by the amplifier was less critical - and such speakers can even seem "overdamped" when driven by a modern amp. However, the woofers in most modern speakers are very heavy, have very powerful motors, and are very tightly coupled. They also tend to have far less loss in the spider and surround... and are often housed in smaller cabinets which provide more spring force as well. In short... they are designed to work with amplifiers that provide lots of electrical damping by sinking that back-EMF-driven current. (That's why a lot of modern speakers tend to sound "sloppy" if connected to tube amps... which provide relatively little damping.) Also bear in mind that, with speakers, the speaker itself is driven in both directions. So, yes, you are providing driving force to drive the speakers in both directions. But you are then also providing braking force in both directions... This braking force is required to brake the speaker when it attempts to overshoot in either direction... But, unlike a motor, the speaker is also "a weight and a spring force"... So, when no drive is applied, it will continue to move, to bounce back and forth at its resonant frequency, because it has stored energy that it needs to dissipate... (This is what you see on those "waterfall plots" where the speaker keeps producing sound at certain frequencies after the signal has stopped.) And the amplifier also applies braking force against that energy by sinking the back-EMF-driven current it generates. With many speakers this is so extreme that it is quite easy to hear... Just tap the woofer gently with your knuckle with nothing connected to the terminals... Then repeat this with the terminals shorted with a paper clip... With many speakers you will hear a big difference. (The paper clip applies "dynamic braking" by sinking that back-EMF-generated current.) Remember that a speaker has a permanent magnet... so it acts exactly like a generator if you "drive it in reverse". And, in this case, the momentum of the speaker mass moving provides the energy to drive it. I come from an industrial motor / control background so I tend to see thing more in that perspective. A speaker is a motor to me although movement is relatively small. You can power it in one direction and allow friction or even spring force to remove the energy required to slow it down. You can power it in one direction and allow friction, spring force, and regeneration to slow it down (EMF). Or, if you need high performance, you power it both forward and in reverse by using a bidirectional power supply. Changing the amount and direction of current in a highly inductive circuit can be greatly enhanced by a voltage twice as high or more to force the current change to comply faster than normal resistive EMF discharge time. An audio push-pull amp does this to some extent, but if the voltage were raised, it would control faster and better by pure electro-magnetic force. Another difference between motor control and audio speakers is that the motor circuits are 'current' regulated and controlled vs the open loop 'voltage' control of audio amps on speakers. And so then there are servo speakers with accompanying amplifiers. The Rhythmik subs, for example. When I build the Open Baffle Dipole Subs from GR Research sometime this year, each of the three 12" drivers for each sub has a second coil which provides feedback to the amplifier. The servo amp has settings to adjust the amount of damping. Seems like such a good idea to be able to optimize the driver with regard to certain properties and then actively manage its movement with the servo system. But it doesn't seem that this method is used much.
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KeithL
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Post by KeithL on Mar 12, 2024 9:38:15 GMT -5
The idea of using feedback and servo-control for a speaker has been around for a long time. I believe the early Velodyne models actually had an accelerometer mounted on the cone... And there have been several that use a separate feedback coil... Unfortunately, in practice, there are a whole bunch of limitations and drawbacks, and not all that many viable benefits. I'm just going to throw out a few here... 1. A feedback coil does NOT sense the position of the driver. It also does NOT technically tell you what the cone, or the surround, or the air in the room, is doing. The feedback it provides is for the velocity of the voice coil through the field produced by the magnet. a. you need to convert that velocity data into the position of the driver b. the accuracy of that data is limited by the consistency of the magnetic field c. it's going to need to be calibrated for the specific coil and magnet you have d. not only is the magnetic field probably not perfectly consistent to begin with but it is modulated by the field generated by the voice coil e. the coil senses the velocity of the voice coil and so cannot account for flexing of the cone f. there are also distinct limitations on how much correction can be applied and how quickly g. you are ASSUMING that the cone is a perfect piston and so the position of every point on the cone can be extrapolated from the position of the voice coil 2. There is a significant amount of delay in your "servo-system". a. the driver is physically heavy, and so has inertia, and the voice coil has significant inductance... b. the movement of the driver is actually the result of the sum of forces applied over some previous time interval - which may be significant (that sum may also include things like spring force that was stored in the air pressure in the cabinet, or momentum of air moving in a port, during a previous driver excursion) c. therefore there is a significant delay in the time between "sensing an error" and "the correction actually being applied to the voice coil". Also, when tuning and adjusting a servo system, a lot relies on your system having extremely predictable behavior... Unfortunately, with a loudspeaker, there are a LOT of variables... For example, the weight of the cone may change when the air is damp, because the cardboard may absorb moisture... And, even if we avoid that, the actual density of the air will change (and so the "moving mass" of the cone and the air it is coupled to will change)... Likewise, on a rear-ported system, changing the distance to the wall will change the port loading... That's a LOT of variables to take into account. As I noted in another post this has generally only been applied to subwoofers... Which are big, heavy, slow moving drivers which really do tend to act more or less like an ideal piston... And there may be a significant benefit to "merely linearizing the movement of the driver itself"... However.... When it comes to midranges and tweeters... Take a look at a few waterfall plots... Or, if you can find them, images of the actual movement of the surface of the driver... To be quite blunt they DO NOT act like perfect pistons and we DO NOT have precise control over their movements... The control we have over the system is much more "casual"... Energy bounces back and forth... And energy leaves the voice coil, travels to the edge, and gets absorbed by the surround... And some of that energy fails to be absorbed and bounces back again... And the material of the cone or dome itself flexes and ripples... And energy from the back of the driver bounces back from the rear enclosure... And the surface of the driver rings at its resonant frequency (or frequencies) due to its mass and spring constants... (And you really cannot correct an unwanted resonance with a servo-mechanism... at least not safely.) I think it would be fair to say that most drivers are designed such that: "when we dump energy into them, some of that energy is radiated to the air in a predictable way, and most of the remainder is dissipated harmlessly somewhere internally". For a typical driver... about 5% of the energy we put in is coupled to the air as output... And the remainder is "burned" somewhere in the various "damping mechanisms"... (hopefully "harmlessly") This is simply not the sort of mechanism that is conducive to "precise servo-control". If you look at the waterfall plot of a typical driver you will see that a significant part of the output occurs AFTER THE SIGNAL HAS STOPPED. That is all energy that has become stored "in the system" and must somehow be dissipated... (It's all "error"... ) Look at something like a motor or solenoid... If it continues moving, after the drive is removed, due to inertia, we can apply a "nice neat braking force" to stop it... since it essentially moves "smoothly in one dimension". Now try to imagine the sort of correction a servo system would need to be able to calculate and apply to "level one of those waterfall plots"... (And remember that the waterfall plot only shows the speaker's output at a single point in space... but the real output varies over an area.) As I mentioned in another post... There are small drivers that are comprised of arrays of individually controlled piezo-electric elements... And, for those drivers, it might be practical to use some sort of servo control on each individual element... I don't see this being practical for drivers that function the way current drivers do... (Except to linearize the motion of the cone itself in very low frequency drivers like subwoofers.) Also, as you noted earlier, very few subwoofer manufacturers do this... And the obvious conclusion is that they fail to see a significant benefit that would outweigh the cost and other "down-sides"... It's not that difficult to produce a subwoofer driver that can deliver relatively linear cone movement using a powerful motor structure, a stiff cone, and a relatively linear surround. It's not clear that a servo-system could actually deliver significantly better performance at a reasonable cost. And, to put it bluntly, there isn't much demand for super-low distortion at extremely low frequencies. Most subs have relatively high THD at very high outputs... and it doesn't seem to be all that important for most applications subs are used for. (Most of the content in that LFE channel is things like explosions... or other sounds with lots of harmonic content... so a little extra THD simply isn't noticed.) True... And, in older loudspeakers, where the motors were weaker, and the coupling much looser, it was generally assumed that the speaker would provide significant mechanical damping and other similar losses. This is why the damping factor provided by the amplifier was less critical - and such speakers can even seem "overdamped" when driven by a modern amp. However, the woofers in most modern speakers are very heavy, have very powerful motors, and are very tightly coupled. They also tend to have far less loss in the spider and surround... and are often housed in smaller cabinets which provide more spring force as well. In short... they are designed to work with amplifiers that provide lots of electrical damping by sinking that back-EMF-driven current. (That's why a lot of modern speakers tend to sound "sloppy" if connected to tube amps... which provide relatively little damping.) Also bear in mind that, with speakers, the speaker itself is driven in both directions. So, yes, you are providing driving force to drive the speakers in both directions. But you are then also providing braking force in both directions... This braking force is required to brake the speaker when it attempts to overshoot in either direction... But, unlike a motor, the speaker is also "a weight and a spring force"... So, when no drive is applied, it will continue to move, to bounce back and forth at its resonant frequency, because it has stored energy that it needs to dissipate... (This is what you see on those "waterfall plots" where the speaker keeps producing sound at certain frequencies after the signal has stopped.) And the amplifier also applies braking force against that energy by sinking the back-EMF-driven current it generates. With many speakers this is so extreme that it is quite easy to hear... Just tap the woofer gently with your knuckle with nothing connected to the terminals... Then repeat this with the terminals shorted with a paper clip... With many speakers you will hear a big difference. (The paper clip applies "dynamic braking" by sinking that back-EMF-generated current.) Remember that a speaker has a permanent magnet... so it acts exactly like a generator if you "drive it in reverse". And, in this case, the momentum of the speaker mass moving provides the energy to drive it. And so then there are servo speakers with accompanying amplifiers. The Rhythmik subs, for example. When I build the Open Baffle Dipole Subs from GR Research sometime this year, each of the three 12" drivers for each sub has a second coil which provides feedback to the amplifier. The servo amp has settings to adjust the amount of damping. Seems like such a good idea to be able to optimize the driver with regard to certain properties and then actively manage its movement with the servo system. But it doesn't seem that this method is used much.
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Post by mgbpuff on Mar 12, 2024 11:33:12 GMT -5
Maybe the feedback device could be a laser distance measuring instrument looking directly at the driver (with high enough response, of course - if that is possible). Closing the loop this way would eliminate all of the variables Keith is mentioning.
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Post by marcl on Mar 12, 2024 12:42:18 GMT -5
The idea of using feedback and servo-control for a speaker has been around for a long time. I believe the early Velodyne models actually had an accelerometer mounted on the cone... And there have been several that use a separate feedback coil... Unfortunately, in practice, there are a whole bunch of limitations and drawbacks, and not all that many viable benefits. I'm just going to throw out a few here... 1. A feedback coil does NOT sense the position of the driver. It also does NOT technically tell you what the cone, or the surround, or the air in the room, is doing. The feedback it provides is for the velocity of the voice coil through the field produced by the magnet. a. you need to convert that velocity data into the position of the driver b. the accuracy of that data is limited by the consistency of the magnetic field c. it's going to need to be calibrated for the specific coil and magnet you have d. not only is the magnetic field probably not perfectly consistent to begin with but it is modulated by the field generated by the voice coil e. the coil senses the velocity of the voice coil and so cannot account for flexing of the cone f. there are also distinct limitations on how much correction can be applied and how quickly g. you are ASSUMING that the cone is a perfect piston and so the position of every point on the cone can be extrapolated from the position of the voice coil 2. There is a significant amount of delay in your "servo-system". a. the driver is physically heavy, and so has inertia, and the voice coil has significant inductance... b. the movement of the driver is actually the result of the sum of forces applied over some previous time interval - which may be significant (that sum may also include things like spring force that was stored in the air pressure in the cabinet, or momentum of air moving in a port, during a previous driver excursion) c. therefore there is a significant delay in the time between "sensing an error" and "the correction actually being applied to the voice coil". Also, when tuning and adjusting a servo system, a lot relies on your system having extremely predictable behavior... Unfortunately, with a loudspeaker, there are a LOT of variables... For example, the weight of the cone may change when the air is damp, because the cardboard may absorb moisture... And, even if we avoid that, the actual density of the air will change (and so the "moving mass" of the cone and the air it is coupled to will change)... Likewise, on a rear-ported system, changing the distance to the wall will change the port loading... That's a LOT of variables to take into account. As I noted in another post this has generally only been applied to subwoofers... Which are big, heavy, slow moving drivers which really do tend to act more or less like an ideal piston... And there may be a significant benefit to "merely linearizing the movement of the driver itself"... However.... When it comes to midranges and tweeters... Take a look at a few waterfall plots... Or, if you can find them, images of the actual movement of the surface of the driver... To be quite blunt they DO NOT act like perfect pistons and we DO NOT have precise control over their movements... The control we have over the system is much more "casual"... Energy bounces back and forth... And energy leaves the voice coil, travels to the edge, and gets absorbed by the surround... And some of that energy fails to be absorbed and bounces back again... And the material of the cone or dome itself flexes and ripples... And energy from the back of the driver bounces back from the rear enclosure... And the surface of the driver rings at its resonant frequency (or frequencies) due to its mass and spring constants... (And you really cannot correct an unwanted resonance with a servo-mechanism... at least not safely.) I think it would be fair to say that most drivers are designed such that: "when we dump energy into them, some of that energy is radiated to the air in a predictable way, and most of the remainder is dissipated harmlessly somewhere internally". For a typical driver... about 5% of the energy we put in is coupled to the air as output... And the remainder is "burned" somewhere in the various "damping mechanisms"... (hopefully "harmlessly") This is simply not the sort of mechanism that is conducive to "precise servo-control". If you look at the waterfall plot of a typical driver you will see that a significant part of the output occurs AFTER THE SIGNAL HAS STOPPED. That is all energy that has become stored "in the system" and must somehow be dissipated... (It's all "error"... ) Look at something like a motor or solenoid... If it continues moving, after the drive is removed, due to inertia, we can apply a "nice neat braking force" to stop it... since it essentially moves "smoothly in one dimension". Now try to imagine the sort of correction a servo system would need to be able to calculate and apply to "level one of those waterfall plots"... (And remember that the waterfall plot only shows the speaker's output at a single point in space... but the real output varies over an area.) As I mentioned in another post... There are small drivers that are comprised of arrays of individually controlled piezo-electric elements... And, for those drivers, it might be practical to use some sort of servo control on each individual element... I don't see this being practical for drivers that function the way current drivers do... (Except to linearize the motion of the cone itself in very low frequency drivers like subwoofers.) Also, as you noted earlier, very few subwoofer manufacturers do this... And the obvious conclusion is that they fail to see a significant benefit that would outweigh the cost and other "down-sides"... It's not that difficult to produce a subwoofer driver that can deliver relatively linear cone movement using a powerful motor structure, a stiff cone, and a relatively linear surround. It's not clear that a servo-system could actually deliver significantly better performance at a reasonable cost. And, to put it bluntly, there isn't much demand for super-low distortion at extremely low frequencies. Most subs have relatively high THD at very high outputs... and it doesn't seem to be all that important for most applications subs are used for. (Most of the content in that LFE channel is things like explosions... or other sounds with lots of harmonic content... so a little extra THD simply isn't noticed.) And so then there are servo speakers with accompanying amplifiers. The Rhythmik subs, for example. When I build the Open Baffle Dipole Subs from GR Research sometime this year, each of the three 12" drivers for each sub has a second coil which provides feedback to the amplifier. The servo amp has settings to adjust the amount of damping. Seems like such a good idea to be able to optimize the driver with regard to certain properties and then actively manage its movement with the servo system. But it doesn't seem that this method is used much. Wow ... as usual you dove in with a ton of context and broad-based pros and cons. And as usual ... way down in there you kind of answered the question with respect to subs, I think: a servo solution could be a benefit for subs due to their size and mass. So I think the broad answer for subs is, there are still lots of variables ... you can't make a generic servo amp and expect it to work with a lot of drivers. Rhythmik subs tend to be favored by Magnepan owners because it seems in their specific implementation with their drivers and amps designed to work together ... they have overcome the pitfalls. Extensive discussion of their technology here .... www.rythmikaudio.com/technology.html
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KeithL
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Post by KeithL on Mar 12, 2024 14:14:37 GMT -5
I took a quick look... and their explanation seems to be pretty thorough... and pretty straightforward. From their description it seems like their driver has a second "sense coil" that is somewhat specialized. They made a point of the fact that "they aren't just using the second voice coil on a dual-voice-coil woofer" and mentioned "thinner wire". I would assume that it's thinner, lighter, probably has more windings, and may be spaced away from the main VC for thermal isolation. (However, from their description, it seems obvious that it's a sense coil, so saying that "it acts like a microphone" is a bit misleading.) Their descriptions of the benefits they were able to achieve also seem quite reasonable... and not at all unexpected. And, yes, it's sort of obvious that both the driver and the amp are custom designed, and their entire system is optimized to work together. The other thing you need to keep in mind is that, with systems of this sort, there is going to inevitably be a lot of "manual fine tuning" and "trial and error". For example, back in the old days, I recall seeing articles about people who designed and built DIY subwoofers using accelerometers for feedback sensing. (And, as they mentioned, a few that used the second voice coil in a dual-voice-coil driver for feedback.) However, designing and building something like that, and then getting all the bugs out, so that it works well, is a MAJOR undertaking... (There are both a lot of things that must be taken into account - and a lot of things that can go wrong.) I have to admit that I haven't heard one of their subs in person (at least not when I was paying attention)... But I've "always heard good things about them"... ................................ Wow ... as usual you dove in with a ton of context and broad-based pros and cons. And as usual ... way down in there you kind of answered the question with respect to subs, I think: a servo solution could be a benefit for subs due to their size and mass. So I think the broad answer for subs is, there are still lots of variables ... you can't make a generic servo amp and expect it to work with a lot of drivers. Rhythmik subs tend to be favored by Magnepan owners because it seems in their specific implementation with their drivers and amps designed to work together ... they have overcome the pitfalls. Extensive discussion of their technology here .... www.rythmikaudio.com/technology.html
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Post by KeithL on Mar 12, 2024 14:46:58 GMT -5
That's a nice idea... but not at all practical... for several reasons. And, no, it will NOT eliminate even most of the variables. [ NOTE that it would eliminate one of the variables quite nicely.] Let's assume that you can measure the location of a single point on the cone instantly and with perfect accuracy. That really still only addresses some of those variables... and not all of those completely. - You STILL have the fact that, since the cone is not perfectly stiff, it will not act like a perfect piston. (So you cannot assume that the entire area of the cone is represented by your single measurement.) - And, even worse, the surround, which makes up a significant part of the area, doesn't act like a piston at all. - And, even with a "closed loop", your ability to correct errors is far from absolute. (There is a limit to the power of your "motor" and to the amount of force that the voice coil and cone can survive.) (The inductance of the voice coil is also a limiting factor because it limits the speed at which you can feed energy into that motor.) (It's also worth mentioning that, since your cone is not perfectly stiff, corrections applied at the voice coil will also "not be applied perfectly to the entire cone".) It's not impossible that the 8-9 dB improvement that Rhythmic claims to be able to achieve with their feedback coil system may actually be approaching the limit of what can be achieved without somehow eliminating those other variables. Incidentally... and just for fun... here's your LASER measurement device... sort-of... www.klippel.de/fileadmin/klippel/Bilder/Know-How/Literature/Papers/Measurement_and_Visualization_Cone_vibration_06.pdfMaybe the feedback device could be a laser distance measuring instrument looking directly at the driver (with high enough response, of course - if that is possible). Closing the loop this way would eliminate all of the variables Keith is mentioning.
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