yes i have it time aligned. I use DSP /no kosher, i know/ so i can delay. In my setup everything is flush so no diffraction problem. I'm talking about different polar characteristic of different tweeters. Even if you setup them that each is ruler flat - each sound very different. 3/4 dome can sound like rear ambient tweeter on some speakers, ribbon sound different etc... In this context it is hard to speak about which is better it is just different. This is why i was interested in your implementation.
Tomas
Hmm, we're talking about very different implementations. My system is powered by a 20-watt direct-heated-triode PP amplifier, with an all-vacuum-tube signal path from the current-output of the Burr-Brown 1704 DAC to the speaker terminals.
I'm using the large RAAL Lazy Ribbon as a supertweeter, and find system integration straightforward. 3rd and 4th-order passive crossovers seem to be doing the job, with a compact impulse response of about 0.5 mSec or a bit less. The AH425 (425 Hz, T = 0.707, optimized for 7 degree exit/entrance angle) is in free air 2" above the top of the bass module, and the RAAL is in free air as well. None of the drivers are requiring in-band equalization or notch filters.
I have no idea how the new loudspeaker would sound with a digital crossover, digital EQ/time delay, and Class AB or Class D transistor amplifiers. The new speaker is designed for people who have direct-heated triode and pentode amplifiers in the 4 to 60-watt power range. (Single-ended 2A3, 300B, 211, & 845, PP Class A 2A3, 300B, 211 & 845, and PP Class AB EL84, EL34, 6L6, KT88, & 6550 amplifiers.)
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Well, here's at least one:
JBL Synthesis CEDIA 2011 Demo Evaluation - Audioholics Home Theater Forums
Well, people can argue about the audibility of diffraction in horns, but as shown in Bjorn's simulation, and confirmed by impulse measurements, it is a real phenomenon, not an audiophile myth.
As for subjective audibility, it probably depends on the listener, with a secondary dependence on the system. Many listeners do not hear time-domain problems in loudspeakers; other listeners can even hear absolute phase (but I'm not one of them). A small minority (possibly as high as 10~15%) can't hear stereophonic phantom images - whenever they hear stereo, all they perceive is two loudspeakers, with no phantom image between the two speakers. People with this perceptual deficit may be inclined to denounce stereo as an elaborate fraud: after all, if they can't hear it, why should anyone else?
The type of music preference makes a difference; many audiophiles never listen to acoustic music performed in real-world acoustical environments (otherwise known as classical music). If all they listen to is studio-created music performed on electronic instruments, there is no real-world reference, just a personal preference for a certain type of sound. Conversely, if a listener loves the sound of acoustical music in a beautiful-sounding hall, they are made aware of the odd colorations of electromechanical reproduction, and will try and minimize the problems they hear with playback.
There's also a system dependence; to my perceptions, at least, there's a noticeable difference in naturalness, vividness of musical timbre, subtlety of detail, and expressiveness when comparing vacuum-tubes to solid-state. Subjectively, I find the best solid-state coarser and less resolving than the best vacuum-tube circuits, and if the front end of the system has subjectively lower resolution, time-domain errors (diffraction, etc.) in loudspeakers are less audible.
My yardstick is naturalness of voice and acoustical instruments in a real acoustic performing space, but not all audiophiles share that.
As for subjective audibility, it probably depends on the listener, with a secondary dependence on the system. Many listeners do not hear time-domain problems in loudspeakers; other listeners can even hear absolute phase (but I'm not one of them). A small minority (possibly as high as 10~15%) can't hear stereophonic phantom images - whenever they hear stereo, all they perceive is two loudspeakers, with no phantom image between the two speakers. People with this perceptual deficit may be inclined to denounce stereo as an elaborate fraud: after all, if they can't hear it, why should anyone else?
The type of music preference makes a difference; many audiophiles never listen to acoustic music performed in real-world acoustical environments (otherwise known as classical music). If all they listen to is studio-created music performed on electronic instruments, there is no real-world reference, just a personal preference for a certain type of sound. Conversely, if a listener loves the sound of acoustical music in a beautiful-sounding hall, they are made aware of the odd colorations of electromechanical reproduction, and will try and minimize the problems they hear with playback.
There's also a system dependence; to my perceptions, at least, there's a noticeable difference in naturalness, vividness of musical timbre, subtlety of detail, and expressiveness when comparing vacuum-tubes to solid-state. Subjectively, I find the best solid-state coarser and less resolving than the best vacuum-tube circuits, and if the front end of the system has subjectively lower resolution, time-domain errors (diffraction, etc.) in loudspeakers are less audible.
My yardstick is naturalness of voice and acoustical instruments in a real acoustic performing space, but not all audiophiles share that.
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I had a first listen in disorganized room with my fiancee running for shelter
Big, tight , bad sound. Impressive and confused would be the word . It's funny how my girl commented on the woofers performance (she's got bat's ears) ..."they do sound little different don't you think" Well, I said... Lynn said the lower ones are diffracting from the floor
I will be parting them out next month.. but in the meantime I will enjoy simple redneck pleasures (Fleetwood Mac playing)
Big, tight , bad sound. Impressive and confused would be the word . It's funny how my girl commented on the woofers performance (she's got bat's ears) ..."they do sound little different don't you think" Well, I said... Lynn said the lower ones are diffracting from the floor
I will be parting them out next month.. but in the meantime I will enjoy simple redneck pleasures (Fleetwood Mac playing)Attachments
Man, those things are BIG! I can see why she's trying to escape!
Don't beat on yourself too hard. WTW's are not easy. I don't care what theory and audiophile-fashion says; fine-tuning the Ariels for an acceptable subjective result was ridiculously time-consuming, and that was with ruler-flat drivers that were a tiny 5.5" across. What measured flat - from many different room locations and the best test equipment on the market at the time - did not sound flat. Getting measurements and subjective listening to agree took many months.
Given my experience with the Ariels, I would never try it with much larger drivers - either a really big WTW or worse, a WMTMW. They might measure flat (although most high-end speakers are far from flat) but in practice sound confused, harsh, and grossly unnatural. Getting loudspeakers of this type of architecture to sound smooth and coherent is very, very difficult. I've yet to hear a truly successful example, and I've heard a lot of loudspeakers.
I am not entirely sure why this is so. Maybe I'm an incompetent designer, or don't know how to measure right, or maybe I hear things that other folks don't. But I think there's really something there; when things sound wrong, they are wrong, and it's up to the designer to be honest about the problem and find out what's going on. That means going outside the literature and examining basic assumptions that might be wrong.
I suspect the assumption of spatial averaging - the usual rationale given for widely spaced woofers - is wrong. Instead of smoothly averaging, the sound becomes incoherent, and the crossover region is a mess - the exact opposite of what a WTW should do. Perhaps a WTW only works when it is spatially compact (small relative to room dimensions), and if the air-load and/or room modes are different for the two bass drivers, pair-matching is destroyed. Maybe.
Looking at the problem more closely, spatial averaging might work, but only at very low frequencies, say, below 80 Hz, where the sense of localization is not very good anyway. At higher frequencies, say 300 Hz and above, the two widely spaced woofers are clearly audible as discrete sources, and the differences in air-load creates small but noticeable phase differences between the two. Maybe.
Don't beat on yourself too hard. WTW's are not easy. I don't care what theory and audiophile-fashion says; fine-tuning the Ariels for an acceptable subjective result was ridiculously time-consuming, and that was with ruler-flat drivers that were a tiny 5.5" across. What measured flat - from many different room locations and the best test equipment on the market at the time - did not sound flat. Getting measurements and subjective listening to agree took many months.
Given my experience with the Ariels, I would never try it with much larger drivers - either a really big WTW or worse, a WMTMW. They might measure flat (although most high-end speakers are far from flat) but in practice sound confused, harsh, and grossly unnatural. Getting loudspeakers of this type of architecture to sound smooth and coherent is very, very difficult. I've yet to hear a truly successful example, and I've heard a lot of loudspeakers.
I am not entirely sure why this is so. Maybe I'm an incompetent designer, or don't know how to measure right, or maybe I hear things that other folks don't. But I think there's really something there; when things sound wrong, they are wrong, and it's up to the designer to be honest about the problem and find out what's going on. That means going outside the literature and examining basic assumptions that might be wrong.
I suspect the assumption of spatial averaging - the usual rationale given for widely spaced woofers - is wrong. Instead of smoothly averaging, the sound becomes incoherent, and the crossover region is a mess - the exact opposite of what a WTW should do. Perhaps a WTW only works when it is spatially compact (small relative to room dimensions), and if the air-load and/or room modes are different for the two bass drivers, pair-matching is destroyed. Maybe.
Looking at the problem more closely, spatial averaging might work, but only at very low frequencies, say, below 80 Hz, where the sense of localization is not very good anyway. At higher frequencies, say 300 Hz and above, the two widely spaced woofers are clearly audible as discrete sources, and the differences in air-load creates small but noticeable phase differences between the two. Maybe.
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This may sound a little arrogant - but what the heck, it is my thread - but the literature is wrong fairly often. We all take it as received wisdom, but if you follow the standard models, the results don't always sound good. Go a little further, dig deeply into the measurements, poke around a little, and you'll find discrepancies that shouldn't be there.
Diffraction is harder to model than you might expect. Damping materials act oddly, and appear to have nonlinear effects that don't fall into any standard model. There's a zero in the response of vented systems, around 1~2 Hz, caused by various box leaks, and that has a strange effect on the linearity of the bass driver as it slowly wobbles back and forth at a 1~2 Hz rate.
Capacitors are genuinely microphonic, and this can be measured by using a 2-foot tube attached to a loudspeaker and beaming sound into a cap with a 9V DC polarizing charge on it. The cap acts like a very low-quality condenser microphone with several big peaks in the 1~5 kHz range. You might think, well, crossover caps don't have a DC polarization, but it's actually worse than that; the first cap of the tweeter circuit has the full voltage of the woofer across it, and LF modulation effectively makes the cap into an AM modulator for the microphonics in the 1~5 kHz range. These are real, measurable effects, not audiophile myths.
To repeat, when things sound wrong, they are wrong. Many designers fall into denial when things sound worse than expected, but that's a signal to dig deeper, instead of denying the evidence of your senses. Sometimes subtler measurements will uncover the true nature of the trouble, but don't expect to find it right away.
The top engineers at ESS took several years to discover why delta-sigma DACs didn't sound quite as good as ladder DACs - and as the inventors of the delta-sigma DAC, they had every incentive to deny what they were hearing and just market the existing DACs all the harder. That's what Philips and Sony did with 44.1/16 CD's, after all - they were marketed as "perfect sound forever" for quite a long time - until the patents ran out, and then of course DSD/SACD became "better than perfect sound for longer than forever".
TomTom has been kind enough to share his results with supertweeters. I have no idea at all why they are not sounding integrated with the mid horn - but then, it's a type of system that I have no experience with. My horns and supertweeter are freestanding and are aligned acoustically; TomTom's system evidently has a common baffle that is shared with the horn-mouth and the supertweeter, and the supertweeter is delayed electronically, presumably with a digital FIR filter with an unknown dithering algorithm and computational bit depth. That's a lot of different variables there.
Diffraction is harder to model than you might expect. Damping materials act oddly, and appear to have nonlinear effects that don't fall into any standard model. There's a zero in the response of vented systems, around 1~2 Hz, caused by various box leaks, and that has a strange effect on the linearity of the bass driver as it slowly wobbles back and forth at a 1~2 Hz rate.
Capacitors are genuinely microphonic, and this can be measured by using a 2-foot tube attached to a loudspeaker and beaming sound into a cap with a 9V DC polarizing charge on it. The cap acts like a very low-quality condenser microphone with several big peaks in the 1~5 kHz range. You might think, well, crossover caps don't have a DC polarization, but it's actually worse than that; the first cap of the tweeter circuit has the full voltage of the woofer across it, and LF modulation effectively makes the cap into an AM modulator for the microphonics in the 1~5 kHz range. These are real, measurable effects, not audiophile myths.
To repeat, when things sound wrong, they are wrong. Many designers fall into denial when things sound worse than expected, but that's a signal to dig deeper, instead of denying the evidence of your senses. Sometimes subtler measurements will uncover the true nature of the trouble, but don't expect to find it right away.
The top engineers at ESS took several years to discover why delta-sigma DACs didn't sound quite as good as ladder DACs - and as the inventors of the delta-sigma DAC, they had every incentive to deny what they were hearing and just market the existing DACs all the harder. That's what Philips and Sony did with 44.1/16 CD's, after all - they were marketed as "perfect sound forever" for quite a long time - until the patents ran out, and then of course DSD/SACD became "better than perfect sound for longer than forever".
TomTom has been kind enough to share his results with supertweeters. I have no idea at all why they are not sounding integrated with the mid horn - but then, it's a type of system that I have no experience with. My horns and supertweeter are freestanding and are aligned acoustically; TomTom's system evidently has a common baffle that is shared with the horn-mouth and the supertweeter, and the supertweeter is delayed electronically, presumably with a digital FIR filter with an unknown dithering algorithm and computational bit depth. That's a lot of different variables there.
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Another possible variable in TomTom's system are the amplifiers for the mid-horn and supertweeter. If they are both Class A, no problem, since distortion in well-designed Class A amplifiers decreases monotonically with lower signal level.
Class AB amplifiers are another story. Distortion increases with decreasing signal level, and if the two amplifiers are fed with a different spectra as a result of the low-level crossover, then the distortion level of the two amplifiers will no longer track each other.
The distortion of Class AB amplifiers is greatest (and most complex) in the zero-crossing region, and if the two amplifier are fed with a different signal spectra, the zero-crossing region will not time-align, since the input signal itself has a different spectra with different zero-crossings. (For one thing, the supertweeter amplifier will have many more zero-crossings than the mid horn amplifier, due to lower overall signal level and greater HF content.) Under dynamic conditions with real-world music, the two amplifiers will have quite different IM spectra, and distortion-tracking is not guaranteed.
I should add that Class AB in vacuum-tube and transistor amplifiers is quite a different thing; in tube amps, the transition region is very broad, covering 20~30 volts, with a gentle transition into the cutoff region for the tube being shut off, while transistors have a on-to-off shutoff in 0.7 volts, and the diode action is very sharp. Transistor amps also have to balance thermal stability against smooth Class AB transitions. For the curious, examining the error signal at the feedback node as signal level is gradually lowered is quite illuminating.
Some transistor amp enthusiasts believe that (Class AB) amplifier distortion is inaudible, but I cannot agree with that premise.
Class AB amplifiers are another story. Distortion increases with decreasing signal level, and if the two amplifiers are fed with a different spectra as a result of the low-level crossover, then the distortion level of the two amplifiers will no longer track each other.
The distortion of Class AB amplifiers is greatest (and most complex) in the zero-crossing region, and if the two amplifier are fed with a different signal spectra, the zero-crossing region will not time-align, since the input signal itself has a different spectra with different zero-crossings. (For one thing, the supertweeter amplifier will have many more zero-crossings than the mid horn amplifier, due to lower overall signal level and greater HF content.) Under dynamic conditions with real-world music, the two amplifiers will have quite different IM spectra, and distortion-tracking is not guaranteed.
I should add that Class AB in vacuum-tube and transistor amplifiers is quite a different thing; in tube amps, the transition region is very broad, covering 20~30 volts, with a gentle transition into the cutoff region for the tube being shut off, while transistors have a on-to-off shutoff in 0.7 volts, and the diode action is very sharp. Transistor amps also have to balance thermal stability against smooth Class AB transitions. For the curious, examining the error signal at the feedback node as signal level is gradually lowered is quite illuminating.
Some transistor amp enthusiasts believe that (Class AB) amplifier distortion is inaudible, but I cannot agree with that premise.
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About litterature being wrong.
Science is newer about reality
Science is about models of reality. This might sound like geeky hair splitting but bear with me.
A good model is a good approximation of reality within some borders set by reality, and by limitations of the model as it models just a limited set of features of reality.
A common model of water flow might be used to describe how water flows thrugh a water pipe of some centimeter and also a river 10 000 times larger but going 10 000 smaller instead and getting into the micrometer size the model is totaly useless.
The fundamental "problem" with sound as percived by us humans is that our logaritmic hearing span over enormous dynamic range going from micro to macro scale and also spanning over a large wavelength range. So the modeling of the physical phenomenons as such is very demanding. Then adding the complexity of human perception of these sound waves, adds several dimentions of difficulties to the mix.
Even trivial things like measuring T/S parmeters are level dependent so of course when the cone behaviour gets more complicated than moving back and fourth as a single unit, things go interesting
Science is newer about reality
Science is about models of reality. This might sound like geeky hair splitting but bear with me.
A good model is a good approximation of reality within some borders set by reality, and by limitations of the model as it models just a limited set of features of reality.
A common model of water flow might be used to describe how water flows thrugh a water pipe of some centimeter and also a river 10 000 times larger but going 10 000 smaller instead and getting into the micrometer size the model is totaly useless.
The fundamental "problem" with sound as percived by us humans is that our logaritmic hearing span over enormous dynamic range going from micro to macro scale and also spanning over a large wavelength range. So the modeling of the physical phenomenons as such is very demanding. Then adding the complexity of human perception of these sound waves, adds several dimentions of difficulties to the mix.
Even trivial things like measuring T/S parmeters are level dependent so of course when the cone behaviour gets more complicated than moving back and fourth as a single unit, things go interesting
Since discussion that i started moves on. I attach picture of my system. I must admit that amp are lousy AB /temporary/. Its build in concrete wall. STW on test was just temporary mounted with lots off cotton wool around to minimize diffraction. System isn't perfect - there is some room for tuning. But WTW pastern is completely without problem - as i say it has to crossed steep enough.
Lynn, i suppose in your case with lazy ribbon - the radiated power /and this is perceived at UHF/ of lazy ribbon is roughly equal of radiated power of Azura at XO, although polar are different shape. Thats why you have less pain with integration.
Just for your info i listened mostly for classical and im also concert-goer. Many thanks for tip with listening to broadband PN long time ago. There are things - very hard measureable but very easy to spot on PN.
And lastly, im not against STW actually i manage proper integration. /But at the end i was capable to get same thing without STW/ It was just not so easy that i hope so and i get many results that was very hard to choose from.
Lynn, i suppose in your case with lazy ribbon - the radiated power /and this is perceived at UHF/ of lazy ribbon is roughly equal of radiated power of Azura at XO, although polar are different shape. Thats why you have less pain with integration.
Just for your info i listened mostly for classical and im also concert-goer. Many thanks for tip with listening to broadband PN long time ago. There are things - very hard measureable but very easy to spot on PN.
And lastly, im not against STW actually i manage proper integration. /But at the end i was capable to get same thing without STW/ It was just not so easy that i hope so and i get many results that was very hard to choose from.
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Jerry Foreman of Precision Audio suggested I use only hermetically sealed Teflon caps for exactly the reason cited. Teflon caps are notoriously microphonic because of the taut winding method used. Regards
You don't have to spend that kind of money for a doubtful gain. Just use an external crossover behind the speaker, something I was doing back at Audionics in 1975. Tie down the caps by wrapping them with soft foam or wool felt and wrapping plastic wire ties around the soft foam or felt (the wire ties will dent the caps otherwise, and you don't want to do that). The plastic ties then go through holes in pegboard or whatever you're building the crossover on. If you want to go further, the microphonic cap can be dunked in a tray of melted beeswax, and let the beeswax slowly harden around the cap with the leads hanging out of the tray.
Along the same lines, the wax caps from Jupiter seem to be the quietest, since they are not made from tightly wrapped plastic and beeswax is not resonant. The tight wrapping seems to be responsible for the microphonics, from what I can tell, and it applies to all wrapped-plastic caps - Teflon, polypropylene, Mylar, etc. Film-and-foil construction is heavier and is mechanically a plastic/metal composite, while the metallization layer of metallized-foil types is so thin it has provides almost no damping for the tightly wound plastic.
Above all else, don't let the caps hang from the connecting wires, where they will be free to vibrate. This is far more important than dumb things like putting little rolling-ball dinguses under a CD player. Although cap microphonics are not as directly audible as vacuum-tube noises, it is responsible for a fair amount of the "cap coloration" we hear - blurring, midrange harshness and grit, lack of dimensionality, etc.
Anyone with a spectrum analyzer can measure cap microphonics. The exciting loudspeaker has to be at least 2 feet away, otherwise the cap will simply pick up the EMF field of the loudspeaker, which defeats the point of the measurement. So use a paper or non-metallic tube to transmit the sound to the cap-under-test.
Polarize the cap with a 9V battery, and have another cap right next to the sound-card or spectrum analyzer input to block the DC (but well away from the loudspeaker). Twist the wires between the cap and analyzer so they don't pick up ambient RFI and noise.
The test setup is the same as the speaker-and-microphone test of a loudspeaker, although you are testing the "microphone" instead of the loudspeaker.
Although the "microphone" output is low, it is not zero, and the spectrum analyzer will display the frequency response of the resulting acoustic pickup of the capacitor. When you see the peaky spectrum on an analyzer, the reason for the coloration will be obvious.
Although I think it's silly to use vibration isolators on solid-state equipment, there is a bit of microphonic pickup from the electrolytic bypass caps which are universally used in solid-state electronics. The power-supply bypass caps are polarized, of course, and will pick up acoustic vibrations and translate them to low-level but quite distorted audio. So not a good idea to put a CD player in front of a compression driver.
I suspect much of the sonics of combining different types of caps - the endless cap-tweaking game that never seems to conclusive - is nothing more than combining different microphonic spectral patterns. Rather than pointless tweaking, better to keep microphonic pickup to the minimum.
Here's my Ariel crossover - not beautiful, but functional, with all caps tied down. The tweeter crossover is on the left side, the midbass on the right. The input switches select between the Marantz home theater amplifier and the Karna high-quality amplifier. The crossover board is elevated from the rug by small foam circles cut from pipe insulation.
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The B fields from air-core inductors spread out several inches - my rule of thumb is 90-degree orientation away from each other and at least 3~6 inches of spacing between inductors. Crossover is external (for microphonic reasons), no ferrous screws are used on the crossover board, and if there is an enclosure for esthetic reasons, it is made of wood or plastic, not aluminum (eddy currents) or steel (nonlinear inductance and unwanted interaction with air-core inductors).
The crossover is spaced off the rug a couple of inches on foam pads to damp vibration and prevent the wires underneath the board (which are the independent star-grounds for each section of the crossover) from snagging on the rug. I keep it at least a foot away from metallic amplifier chassis, or other large bits of metal.
Yes, I have wire preferences, but they're pretty simple: industrial Litz wire with cotton-cloth covering. Litz wire can be tinned and soldered with a solder-pot. If 2 feet of wire sounds different - in any way - than 20 feet, you're using the wrong speaker wire. Likewise, if the wire has to "burn in" for days or weeks, that's the plastic dielectric slowly losing the DC polarization charge that was created by manufacturing stresses. Ditch the plastic and get rid of burn-in hassles for good. If you're worried about triboelectric effects and self-motion from induced currents, better to let the wire move free than contain it in a high-Q resonant plastic straightjacket. Think about it: which is more resonant, tightly wrapped plastic (just like a plastic cap) or dead-soft copper in a Litz-wire bundle in a loose cotton sleeve?
One nice thing about Litz wire is there's no strand-to-strand current flow, thanks to the enamel coating of each strand. So even if the wire moves around a bit, it doesn't matter, since there's no strand-to-strand current flow and associated distortion from copper-oxide corrosion on the surface of each strand.
The crossover is spaced off the rug a couple of inches on foam pads to damp vibration and prevent the wires underneath the board (which are the independent star-grounds for each section of the crossover) from snagging on the rug. I keep it at least a foot away from metallic amplifier chassis, or other large bits of metal.
Yes, I have wire preferences, but they're pretty simple: industrial Litz wire with cotton-cloth covering. Litz wire can be tinned and soldered with a solder-pot. If 2 feet of wire sounds different - in any way - than 20 feet, you're using the wrong speaker wire. Likewise, if the wire has to "burn in" for days or weeks, that's the plastic dielectric slowly losing the DC polarization charge that was created by manufacturing stresses. Ditch the plastic and get rid of burn-in hassles for good. If you're worried about triboelectric effects and self-motion from induced currents, better to let the wire move free than contain it in a high-Q resonant plastic straightjacket. Think about it: which is more resonant, tightly wrapped plastic (just like a plastic cap) or dead-soft copper in a Litz-wire bundle in a loose cotton sleeve?
One nice thing about Litz wire is there's no strand-to-strand current flow, thanks to the enamel coating of each strand. So even if the wire moves around a bit, it doesn't matter, since there's no strand-to-strand current flow and associated distortion from copper-oxide corrosion on the surface of each strand.
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It's fairly simple to validate your orientation, too. I just excite coil 1 with a low AC signal from my function generator, and coil 2 is connected to the scope input. As you move coil 1 (or 2, depending on your preference) and change orientation and distance, you can clearly see which condition provides for the lowest pickup.
The final plus on the side of type 2 Litz wire, which is what you have on hand, is what is called "proximity effect". Basically, to induce a 60 Hz power grid signal, you have to drape the cables across a power transformer. Power cords are ignored, nylon carpets are ignored, RFI is ignored until gigahertz, it even works perfectly in low signal level interconnects as long as three meters with no shielding.
Just don't introduce it to your cat, they cannot help but chew on it....
Bud
Just don't introduce it to your cat, they cannot help but chew on it....
Bud
