The Xenover

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Part one of two

So, what is an active crossover? The short answer is that it’s an electronic filter designed to divide frequencies into ranges suitable for tweeters, midranges, woofers, and subwoofers. Yes, and supertweeters, too…I know you’re out there.

In an ideal world we wouldn’t need filters. You’d only need one driver. Its frequency response would go from DC to light and it would get earsplittingly loud from a single watt. In the here and now, we are faced with drivers that only cover a few octaves well and need copious amounts of power to achieve reasonable volumes. Some way to combine drivers would be nice—using the extended top of one with the deep bass of another. It can be done, of course. All you have to do is assign the deep-ranging frequencies to the woofer, and the high frequencies to the tweeter. Why not let each driver carry all the music; just operate the woofer and the tweeter together in parallel or series, each playing the entire musical spectrum? Because it wouldn’t sound good, and because tweeters can’t take the sort of abuse they would suffer if asked to play organ pedal notes. So we need to separate the high frequencies from the low frequencies and send them to their respective drivers.

Most speakers use passive crossovers. They are comprised of resistors, capacitors, and inductors, usually mounted within the speakers themselves. They contain no active devices, require no external power, and, barring the occasional tweeter level control, don’t need adjustment.

An active crossover performs the same function, but does it further upstream. You’ll need at least two amplifiers, which leads us to the only real downside of multi-amping—cost. Amplifiers cost money. However, at this point enough people have built Alephs, X-variants, and whatnot that there are spare amplifiers sitting on the shelf gathering dust. An active crossover would give folks an excuse to use all that excess hardware.

Once you get past the price, it’s pretty much uphill from there. Multi-amp systems have numerous benefits. Passive crossovers are inefficient. Not all the power that goes in gets to the driver. A passive crossover burns off valuable wattage as heat, gaining you nothing in return. An active crossover loses signal too, but it does so earlier in the amplification process, when gain is cheap. Another benefit is increased damping factor. The damping factor is a ratio; divide the impedance of the driver by the output impedance of the amplifier. All things being equal, a bigger number is better. Basic Newtonian physics: An object in motion tends to remain in motion unless acted upon by an outside force. Start a woofer cone moving forward and it will tend to keep going, even though it’s time to turn around and head the other way. The driver’s suspension and some cabinet designs help oppose the wayward driver’s flight, but they could use some help. Enter another law of physics: For every action, there is an equal and opposite reaction. There is an electronic analog to this that we can use to put the brakes on. First, you have to stretch your imagination and look at the driver as a microphone. Vibration of the cone will produce electric current at the speaker terminals. If you short the driver’s terminals together, the cone suddenly finds itself faced with the current it just created, and in one of those cosmically cool coincidences, the current comes back into the driver via the short and immediately tries to push the driver the opposite direction from the way it’s moving, thus bringing it to a halt much more quickly. So what’s that got to do with amplifiers? Pretend you’re a driver, looking up through the speaker cable into the back of the amplifier. There’s a resistance there—well, actually an impedance—that varies from amplifier to amplifier. It’s generally pretty low, and that’s good. A passive crossover generally has a number of components in it, some of which are effectively increasing the impedance seen by the driver as it looks back up towards the amplifier. This lowers the damping factor considerably. By getting rid of the passive crossover, the driver can get all the benefit of that low Z output—which then serves as the ‘short’ that allows the speaker to put on the brakes. Another benefit to multi-amping is the fact that good parts cost money, and higher voltage, higher wattage parts cost exponentially more money. For the cost of a single decent high voltage cap to go into a passive crossover, you can buy an entire active crossover’s worth of small caps.

So now you’re convinced, right? You want to do the multi-amp thing and put all those Zens and Alephs and JLHs and single-ended tube amps to work. What’s next? Well, you need a topology. There are lots of ways to build an active crossover. Two of the most common are Sallen-Key (named after the guys who developed it), and multiple feedback. So which one is best? That’s a trick question. You’re seeking an absolute answer to a relative question. If you phrase it ‘Which one is better?’ you’ll be closer to getting an answer you can live with, but that still begs the question of,"better for what?" Multiple feedback filters have a clear advantage in certain areas. For instance, they’re easy to design for high Q and they’re adaptable for balanced signals. High Q is rarely needed in audio electronics, but the balanced feature might be nice to have. So what’s Sallen-Key got going for it? Well, it’s capable of relatively easy adjustments and, under the right circumstances, it simplifies parts selection. Hmmm. What to do? We could go either way, but ease of adjustment is a powerful thing to have in our favor, so we’ll reluctantly drop the ability to filter a balanced signal and go with the Sallen-Key.

With the Sallen-Key topology in mind, we find that we need an active component of some sort. It can be as simple as a follower, or so complicated that a tribe of EE’s get lost without road maps. The follower is seductive due to its simplicity, but some gain would be nice, so I’ve drawn the schematics with an opamp. Which opamp? Any opamp. You can build yourself a discrete opamp like the ones that Nelson Pass outlined in his paper on DIY opamps on the www.passdiy.com website, or you can buy a simple chip opamp at Radio Shack if you’re desperate. Try not to get that desperate.

Next comes the pesky question of what to do about the other parts. While it’s possible to build an active filter with inductors, we won’t need them. Resistors and caps are all the Sallen-Key topology requires. Parts quality? How good do you want it to sound? I use the Vishay/Dale 1% resistors from Mouser. Do you need to buy 1% resistors? Not really, but they’ll give you finer control over your crossover points. For caps, I lean towards polystyrene. Again, Mouser has them in a decent selection of values. If you want to use polypropylene or silver/mica or whatever, that’s fine. The circuit won’t care. Just try to have a reasonable match between the two channels, as odd things can happen to the image if one channel’s crossover point is higher than the other’s.

I’m using a slightly strange system of notation on the schematic. The reason being that the parts nomenclature goes along with what the part does, not its position on the schematic or due to any relation to the math behind the scenes. The first set of resistors and caps the signal sees on the way in define the second pole. The second set define the first pole. What’s a pole? A pole is a multiple of 6 dB/oct. A one pole filter is 6 dB/oct. A two pole filter is 12 dB/oct. Three poles equals 18 dB/oct., and so forth. Sorry, they come in sets of six. That’s just the way the universe works. If you want to go for an in-between number, you’re shading off into shelving networks and such and that’s a whole ‘nother ball of wax. Maybe some other day. In any event, C1 will be the cap for pole 1, C2 for pole 2, R1 for pole 1, and R2 for pole 2. I don’t show it on the schematics, but you can use jumpers to cut out pole 2 entirely and make a two pole filter a single pole filter. Purely selfish of me to name things in such an odd way, but it’ll save me from having to carry around a schematic all the time. If we’re talking about R2, I’ll immediately know that it’s the resistor that helps set the frequency for the second pole of the filter.

While there’s nothing patented about this circuit, I’d like to thank Nelson Pass for two ideas which I lifted wholesale from his commercial crossover, the XVR1. I’ve been building crossovers for quite some time, and though some of them had selectable frequencies, I always hard-wired the Q. The idea of selectable Q tickled me so much that I dropped what I was doing and redid everything so as to have the ability to set the Q. Neat. The other thing is a purely practical matter. Headers. You know…those little jumper things they use in computers to set configurations. Cheap, zillions of positions, gold-plated. Love it. So when’s the last time you priced a ten-position switch of decent quality? I betcha it cost more than 39 cents. Mind you, if you want to use Elma or Electroswitch or Alps or whatever’s popular this month, they’ll do just fine. Better than fine. But they’re likely to cost as much as all the other parts put together. For frequency selection you don’t need 100,000 cycle switches. You’ll be working things hard if you change frequencies as much as a hundred times during the life of the crossover. Headers are your friends. Particularly if you don’t have an infinite R&D budget.

And the name of the circuit? Well, remember that X is pronounced as Z, as in xylophone. Also, xeno comes from the Greek language, meaning strange or alien. To some people, this crossover may seem a little strange. So is it Xen-over or Xeno-ver? I’ll leave that up to you.

I’m nearing the limit for length on posts, so I’m going to break here and start another chapter.

Grey
 

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Part two of two

Frequency selection:
The crossover frequency is selected by the values of R1, R2, C1, and C2. I’ve drawn the schematics to show a single cap value and three resistor values for each pole. Simply jumper one resistor from the R1 bank and one resistor from the R2 bank. You’re done. You can easily set it up for more than one cap and as many resistors as you can afford to buy. If you want real flexibility in frequency selection, use a pot instead of fixed resistors.

Now, let’s get down to the nitty-gritty. How do you decide what values to use for your resistors and capacitors? There’s only one formula you need to know:

F=1/(2*PI*R*C)
where
F=frequency
R=resistance
C=capacitance in Farads

Generally, you’ll want to put the same values in the R1 bank as you put in the R2 bank. For the purposes of the schematic, I arbitrarily chose a nominal crossover frequency of 500Hz, then bracketed it with values at +-10%. You can change the crossover frequency in either of two ways. You can crank the handle on the formula until the answer pops out, or you can simply scale the parts values in the schematic. It’s an inverse linear relationship. If you want a nominal crossover frequency of 50Hz, then that’s one-tenth of 500Hz. You then increase either the cap or the resistor values by a factor of ten. Since 681k is going to sound like a really high value to a solid state person (tube folks sprinkle 681k resistors on their breakfast cereal—doesn’t bother them a bit), you might be happier going with a 47000pF cap instead.

The idea of selecting the poles separately is a little scary. Most people, most of the time, are going to want to switch them together. That is, if you select resistor R1b, then you’re going to want to select resistor R2b at the same time. This will give you a nice, conventional crossover slope. But if you like to walk on the wild side, you can choose different values. That will give you a compound slope. A compound slope is one that starts at one rate, but changes to another, steeper slope later on. If the values you select are near each other, you won’t notice much difference, but if you select frequencies two or three octaves apart, you can get some really interesting curves. Use this feature to equalize drivers that have a rising response near the end of their usable frequency range (we’ll deal with falling response when we get to Q). If you want to get a two pole rotary switch and switch both poles together, that’ll be just great. The separate pole adjustment thing just sort of falls out when you go with headers for frequency selection. There’s no law that says you have to use it.

Good caps are generally more expensive than resistors, so I set the circuit up with a minimum number of caps and used resistors for frequency selection. If you happen to have a box full of caps you’re itching to use, feel free to reconfigure the circuit to have switchable cap values. For more versatility, use lots of each. If you really want to go to the limit, jumper more than one value at a time. This will place the selections in parallel, and may take a little calculation to get the desired result, but jumpers are only a few cents apiece, and you can fiddle to your heart’s content.

How do you know which slope to choose? A good place to start is to duplicate what the manufacturer was using in their passive crossover. Same frequency, same slope. It’s often in the specifications. Once you get used to that, you can try varying things a bit. If you’re starting from scratch, building a DIY speaker, you’re going to have to wing it. There are too many driver choices to try to give guidelines here, although you generally won’t want to go much below about 2kHz with most tweeters; 3kHz is safer still. If you don’t have something specific in mind, stick with a 12 dB/oct slope until you get further along in the design process. Yes, there are people who are passionate about first order crossovers, and for every one of them, there’s a Linkwitz-Riley fan who’s just as rabid about 24 dB/oct slopes. 12 dB/oct is a good compromise to start with. You can always change it later.

First order slopes are easy and the Sallen-Key topology does a good job of handling second order, but what do you do if you want a third or fourth order slope? Add two stages together. A 6 dB/oct stage adds to a 12 dB/oct stage to yield 18 dB/oct, just as you might expect. Likewise, two 12’s yield a 24 dB/oct slope. Crossover slopes above fourth order are rare in audio work.

Not all 12 dB/oct slopes are created equal, nor are 18’s or 24’s or any of the higher order slopes. Only 6 dB/oct slopes sidestep the question of Q. Q is a measure of how "peaky" or "droopy" a crossover slope is at its cutoff frequency. Q is best illustrated with graphs, which I can’t easily generate and post. Look at the final page of the owner’s manual for the Pass Labs XVR1 (available online in PDF format at www.passlabs.com) and you’ll find graphs showing the effect of Q on the slope of the crossover. Nelson has three settings for Q on the XVR1: Low, Medium, and High. If my scribblings are correct, these correspond to Qs of .5, .63, and 1. Note that Q doesn’t have a named unit, like Ohms or Farads, it’s just a number. The schematic shows three crossover selections, .577, .707, and 1. The lowest setting, .577, has a name: Bessel. Likewise, the middle selection, .707, is called Butterworth. These are merely convenient labels for specific points on a continuous line. There’s no reason you can’t design a crossover with a Q of .8153 or any other figure that you might need, although it’s unusual to need anything below .5 or above 2. The Bessel configuration’s claim to fame is that it has the least phase shift. Butterworth has the flattest passband response. The .63 figure that Nelson appears to use for the XVR1 is a good compromise between the two, allowing for flat response with little phase shift. A Q of 1 is useful for bumping up a sagging frequency response curve right before the rolloff. It will provide a boost of 1 dB. Higher boosts are possible; a Q of 2, for instance, will provide 6 dB of boost, but this sort of trick is best used in moderation as it can take a considerable amount of amplifier power and be hard on the driver. Approach equalization with caution. Like fire, it is a good servant, but a bad master.

Suppose you want a proper Butterworth 18 dB/oct filter. How would you go about putting one together? Choose a crossover frequency. Then calculate convenient resistor and capacitor values for that frequency. You’ll need a 6 dB/oct filter and a 12 dB/oct filter. In principle, you can put them in either order, but if you put the 6 first, you’ll get a slight advantage in noise reduction. The last thing to do is to set the 12 dB/oct filter section for a Q of 1. Why 1, if Butterworth is supposed to be .707? Because the 6 dB/oct filter is a bit droopy and the slight boost from the 12’s Q of 1 offsets this and the two work hand in hand to create a good rolloff characteristic. There are extensive tables of ratios to be used for setting filters to various desired Qs, and eventually this thread will probably be filled with scads of suggestions for various filters. It’s a tedious job to transcribe all that information, so I’ll leave it for later, as I’m running short on time.

I’ll close with one final note on how to set Q should you want something other than the ratios that I put in the example schematic. Q is determined entirely by the ratio of Rfb and RQ. That’s one of the benefits of using this filter configuration. It doesn’t really matter whether you change Rfb, RQ, or both. In the final analysis, only the ratio between them matters. In my case, both the schematic and my PCB layout worked best when I left Rfb constant and varied RQ. The formula for determining Q is:

Q=1/(3-(1+(Rfb/RQ)))

The main caveat here is that you’ll want to stick to low Q values. If you start pushing the circuit too hard, it gets really, really sensitive to parts values and you’re going to need to go to very tight tolerance parts in order to get predictable performance. For most purposes, audio filters don’t need to go above a Q of 2 or 3, so this shouldn’t be a problem in the real world.

Not all of this is as clear as I’d like, but one of the nice things about an interactive format like this is that we can hammer at it until (almost) everyone is satisfied. I still haven’t covered phase relationships, amp selection, and power requirements, but that’ll have to wait. I’m going to go pull some gear and go camping.

Grey
 

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diyAudio Editor
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Grey, This is just what I needed. A kindergarten chat about the subject. Lots of us know how hard it is to write clearly, so your efforts are appreciated. Maybe eventually it can go in the wiki
as part of a basic primer aas you had mentioned in the past.

I'm using a balanced preamp and amp, but the CD into the pre is unbalanced. So, do I need to put the crossover before the preamp(s) and have 2 preamps?

I say we call it the Rollover.

If you agree, it's up to Dave to change the posts till now. It wouldn't be hard at this point!

Nelson, have you surrendered to the High/Lo Pass name for your future effort?
 
Grey, thank you very much.

You have a wonderful way of explaining things. Sometimes it is needed just to have this kind of explanation to finally "get" something that you have anyway tought about and studied for quite a while.

Just for curiousity, what king of solution are you using in place of the "opamp" and if you have tried many, which seems to give the best results in your mind?

Ergo
 
Unfortunately I do not know enough about electronics to agree or disagree with Uli's comments, however I do have a few burning questions (BTW thanks for going over the basics!!):

1: The crossover looks fascinating due to its simplicity. However, lets say a person has built a dual-mono, all balanced BOSOZ, and hopes to preserve the entire signal chain as balanced. With this in mind, could that person build two identical crossovers as described above, and use both to process the two halves of the balanced signal??? Just think how cool it would be to have 4 rows of "precision ganged" jumpers!

2: (perhaps a little of subject - sorry) I have previously read that a balanced system rejects noise by eliminating elements not common to both signals, the regular and the inverse. I further read that "this finally happens when the two halves of the signal meet at the speaker terminals". Please help me comprehend how this actually happens. I come from the computer science side of things, and this would make perfect sense if there was some intelligent piece of hardware that filtered noise not common to both signals. But there seems to be no circuitry for that, and it happens completely due to the laws of physics. Some insight would be most helpful.

Thanks again for the great effort!!!
 
amo,
1) Yes, in principle, you can handle each phase of the signal this way, but it's not a truly balanced circuit. It would be a good idea to run everything through a differential afterwords to clean up the accumulated garbage, small though it may be.
2) Look at a differential. There are two gain devices in a sort of a Y shape, with the bottom of the Y being a resistor or current source. If you put a balanced input into a differential, you'll get a balanced signal out--minus anything that was common mode. When you say that something is common mode, you're saying that it shows up the same in both the + and - phases of the signal; same amplitude, same phase, same everything. An example of a common mode signal might be hum due to the signal passing near a transformer.
What happens is this: Let's say a signal goes in the left half of the differential. It's amplified, and in the process the gain device (whether tube, bipolar, or FET) draws current from the base of the Y. At the same time, the other phase of the balanced signal is being amplified by the other half of the differential, but it's going in the opposite direction--if the left half is going positive, then the right half is going negative. This works wonderfully, because as the left half of the differential pulls more current, the right half is using less, then they reverse and as the right half takes more, the left half is releasing exactly the same amount. It's a marvelous balancing act, like well matched, experienced dancers whirling across the floor. It looks effortless.
If a common mode signal is presented to the two inputs they both try to pull or push current at the same time. It's as though two kids on a see-saw both decided to go up at the same time. It doesn't work. The way a see-saw is built means that in order for one kid to go up, the other must go down. Got to. Period. If they both try to go up, nothing happens. The same thing happens in a differential. If you put in a common mode signal, nothing comes out. Well, almost nothing. In the real world, the parts aren't perfectly matched, or the current source doesn't have infinite impedance, etc. and a tiny amount leaks through anyway. This imperfection is rated using a specification called Common Mode Rejection Ratio (aka CMRR--lots less typing). More CMRR is good. It means that the differential is functioning closer to the ideal. It won't ever quite get there, but you can sneak up on it by tweaking this and that.
To have the common mode signal 'cancel at the speaker' you're pretty much going to have to have a balanced amp, e.g. a bridged amp. Nelson's X amps are bridged. The Aleph-X is bridged, but normal Alephs are not. A bridged amp has two halves that amplify separate phases of the signal (even though the circuit may not be based on a differential), then present them to the two ends of the speaker. Pretty much the same thing happens as with the differential. If both sides go positive at the same time, the speaker sees nothing, so common mode stuff is lost in the wash.
If all else fails, you sometimes have to sit down and draw it out on a piece of paper and watch the current go this way and that. Hint: The laws of thermodynamics apply, even though they have different names in electronics. You can't get current from nowhere, and you can't just make it 'go away.'
I'm not quite clear on where Fred was going with his post. If it's meant as a reply to your questions, it seems to be pretty foggy.

Grey
 
So two crossovers + differential = balanced circuit? or do I add something else? I do not mean to sound over obsessed with being balanced, but I want to complete this experiment non the less. My source has a balanced digital out (LynxTwo) and I was intrigued by Nelson's comments about the possibility of running an all balanced DAC using at least 4 separate DAC chips... This is definitely on my to do list. I feel the all balanced BOSOZ is also something I can accomplish with my limited knowledge. The amps is a different story... I think 6 balanced/bridged single ended class A monoblocks (for a 3 way) may be a little much for a beginner. I can see my self getting bogged down.
Ok, couple more questions please: Lets say I choose not to build six amps, and instead decide to use 12 cheap-o units (temporarily I hope), and bridge them together to have 6 balanced pairs. What is the best way to accomplish this? Just have noise cancel at speaker terminals or add a differential network?
Finally- Where can I find more good information on building a differential network?
Thank you very much for your help! This is of tremendous help, as I have found my self daydreaming for hours about balanced active crossovers and CMRR... Is this some kind of a disease? Is there a helpline?
 
As usual, I'm short on time. Came back from camping, walked into work and found that the power had failed. You don't know headaches until you've had a power outage in a major computer installation. Life's been crazy ever since. I'm going to try to take things in order, then post a quick schematic. Anything else I'll try to get to later--let me know if I miss something.
There's no reason why you couldn't put a crossover between the source and the preamp, but it makes life unnecessarily difficult. You'd have to adjust two volume knobs every time you wanted things louder or softer. Typically, the order is source>preamp>crossover>amps>speakers.
Variac,
There's no law that says that you can't use balanced and unbalanced amps side-by-side. I'm assuming that you want to use the SOZ for the high frequencies and the Hafler for the lows, right? Sounds like a winner to me. You can use either the + or - phase to drive the Hafler, as suits.
Generating a balanced signal from an unbalanced one is trivial. Any differential with a half-decent current source under the tail will do the trick. The way Nelson appears to do it in the XVR1 is to receive a balanced input, toss one phase, do the filtering, then recreate the missing phase on the way out. Works fine.
For those who were wondering, yes I fiddled with X-opamps for a while. Got probably five or six variations on the idea scratched out on paper around here somewhere. Decided to go with the single-ended opamp. By all means, if you want to tilt at that particular windmill, be my guest.
amo,
Make sure those twelve cheap-o's are pretty well matched. The bigger problem is that bridging an amp means that each half of the amp sees half of the load--an 8 ohm load appears as a 4 ohm load to one or your 'cheap-o's'. If you're running a 4 ohm speaker, then each amp will see a 2 ohm load. That might not work so well. Just make sure your amps can handle the load.
There's not a requirement to run through a differential, that's just a tidy way to sift out some of the junk. You could just run straight from the two opposite-phased crossover circuits into the amps. The choice is yours.
amo and Thomas,
A balanced filter will pretty much require you to go to what's known as a multiple feedback topology. Balanced operation of a multiple feedback filter is simple, you just duplicate the filter on both inputs of the opamp. There's nothing tricky about it. The downside is that you lose the ability to easily adjust things the way you can with the S-K topology. As for references you can get to on the web, Texas Instruments has a lot of information on multiple feedback filters on their website. A great deal of it is redundant, but what you need to know is there. Just bear in mind that you're going to need a much better idea of the frequencies and Qs of your crossover points going into a multiple feedback filter than with an S-K because you're losing flexibility.
Finally, I'm including a schematic for an opamp that is probably similar to the one Nelson is using in the XVR1. There's one interesting thing in that the manual for the unit states that it can output 30Vrms. That would imply something like +-45V rails, which I find unlikely, since Nelson is probably using the 2SK389 for the JFET front end and it's a 50V part. Nelson isn't noted for running parts at 90% or their rated specs. I find it much more likely that he's running 32V rails (not coincidentally, the range he uses in the DIY opamp paper), which would then lead to 30V peak-to-peak output as opposed to RMS.
As for a complete reverse-engineering of the XVR1, I don't feel that it's necessary. If you've got a usable opamp and the filter section, you're there, functionally. At this point, the only thing remaining would be the chip resistors, but I prefer the flexibility of discrete parts. I'm sure that with enough head scratching the resistor network could be unravelled, but using a single resistor (per pole) to set the frequency is pretty minimalist, albeit at slightly higher parts cost. Face it, we're talking 15-20 cent parts, here, and we're not contrained by board size. It strikes me as a non-issue.
More later.

Grey
 

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diyAudio Retiree
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As usual, I'm short on time.

........ and even shorter on basic electronic knowledge. For a sine wave, the rms value is 0.707 times the peak value. Looking at how far the peak value the signal swing with respect to ground for each output. The specified 15 volts rms would give a peak value of 1/0.707 x 15 or 21.2 volts peak. Giving a little headroom for the rails so the signal is not clipping would seem to give a value of about 24 volts for the positive and -24 volts for the negative rail. I have also seen Erno Borbely use these values for supplies with jfet circuits as well. Your statement "which would then lead to 30V peak-to-peak output as opposed to RMS" seems to imply that Pass Labs is pretty careless with their documentation.

http://whatis.techtarget.com/definition/0,,sid9_gci213722,00.html

http://www.borbelyaudio.com/ae599bor.pdf
http://www.borbelyaudio.com/ae699bor.pdf

Statements like:

"Finally, I'm including a schematic for an opamp that is probably similar to the one Nelson is using in the XVR1."

"As for a complete reverse-engineering of the XVR1, I don't feel that it's necessary. If you've got a usable opamp and the filter section, you're there, functionally. At this point, the only thing remaining would be the chip resistors, but I prefer the flexibility of discrete parts. I'm sure that with enough head scratching the resistor network could be unravelled, but using a single resistor (per pole) to set the frequency is pretty minimalist, albeit at slightly higher parts cost. Face it, we're talking 15-20 cent parts, here, and we're not contrained by board size. It strikes me as a non-issue."

"There's not a requirement to run through a differential, that's just a tidy way to sift out some of the junk. You could just run straight from the two opposite-phased crossover circuits into the amps. The choice is yours."

"Balanced operation of a multiple feedback filter is simple, you just duplicate the filter on both inputs of the opamp. There's nothing tricky about it."

Leave me howling with laughter at your presumptuousness. Idle spectulation is a lot easier than actually providing a schematic of an actual design isn't it? I would even settle for a schematic from someone else to back up evn one or two of your claims. I take it that you won't mind if I run some simulation of your descrete op amp circuit to see how speculatory that is?
May I suggest you do some of your design work on the computer since the designs you commit to paper seem to vanish quite often. You don't have a dog that possibly eats paper do you? The dog ate my homework seems like an excuse that might work for you......
 
Grey,

Thanks for your post and response to specific queries, your relaxed approach is preferred for the average diyer punter.

Actually your articles read a lot like Mr Pass articles.

Your schematic is interesting. Can you elaborate on the theory of operation and why you have selected certain valves?

I have ordered some parts not unlike those in Mr Pass diy opamp article to build up some simple designs to learn how they operate for tutorial purposes.

For those that wish to understand the theory and practical aspects this is a good ideas.

Regarding the max output swing which is trivial, I recall 15 volts SE and 30 balanced from the XVR1 manual.

(I read this can be done in both the Passdiy and Borbely articles)

I see also from reading the Borbely article part 2, there is a nice little buffer , a source follower that would appear to work also if you are a minimalist. But would require different filtering agian this is a minimalist approach. See Borbely's paper Part 2 page 19 and the Pass Phase Coherent Crossover article).





macka
 

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Grey:

Thank you for your response to my question. You have confirmed what I suspected had to be done, but, was unable to find a reference for. I now understand what needs to be done to design an all balanced crossover and what the draw back will be.

I believe that the advantages of an all balanced system out weight the ability to make changes after the system is built. However, for a three way speaker, I think a bread boarded single ended S-K approach will be needed to develop the speaker crossover points and related parameters.

Thanks for the fine work in what turns out to be a less then pleasant environment. I am looking forward to more of it.
 
For the Designers and Builders: :)

The Texas Instrument site has everything anyone would want to know about audio filters. It covers fully differential nicely. (Some minor details are left to the student)

Go to www.texasinstruments.com. Try Knowledgebase, Analog and Mixed Signal, then Op Amp Filters under the frequently asked questions.


For the Philosophers and Poets: :dodgy:

Go bit the bird, lads; save our fellow students for something more worthwhile, like buying the drinks. :drink:

Happy Thanksgiving Everyone:
 
Actually, the discrepancy that has Fred in such a dither is that I was remembering the balanced output voltage, as opposed to the unbalanced. The balanced output of the XVR1 is specified as 30V, and the unbalanced as 15V. 24 or 25V rails sound good to me. I was holding Olivia in my lap this morning as I was putting that post together and between her bouncing up and down and attempting to play patty-cake on the keyboard, I'm pleased to have gotten the post done at all. Yes, Fred, you can simulate to your heart's content.
All this anger and emotion over such a simple thing...jeez.
Now, let me see if I can address Fred's other concerns, before he works himself into such a froth that he has a heart attack. I'd hate to have that on my conscience.
-- I gather that he's concerned about my statement that the S-K topology isn't appropriate for a balanced circuit, whereas a multiple feedback filter is. Since Fred seems to prefer Revealed Truth from an Authority, let's ask Texas Instruments. I trust that they will serve as a suitable Authority for our current purposes. I quote from "More Filter Design on a Budget," SLOA096: "Fully differential modification is easily accomplished by duplicating the feedback path for the MFB topology. It is not possible for Sallen-Key topology." MFB stands for multiple feedback, in this case. They're talking about low pass filters, but the phrasing for high pass filters is identical.
--Fred asks for a schematic. Very well. A schematic can be found on the first page of Cirrus Logic's AN48, where a multiple feedback low pass circuit is shown in balanced configuration. I quote from the text below it: "The 2-Pole Low-Pass Filter with Differential Input is easily designed using the design equations for the multiple-feedback low-pass filter. Also, notice the similarities between Figure 1 [the balanced filter] and Figure 3 [a simple differential amp with balanced inputs]. The differential input function is accomplished by simply duplicating the component values generated in the filter design." I don't know how that sounds to Fred, but it seems clear enough to me.
--Fred seems concerned about my dog. Well, that's an easy one. I have no dog. I prefer deer, raccoons, and such to be able to wander through my yard. A dog would give chase and drive them off. Words cannot express the depth of pain that I feel on hearing that Fred is upset because my notes aren't at my elbow for immediate access. Why, I do believe my discomfort might even go as deep as the thickness of a single sheet of paper.
--I didn't start the thread with the express intention of dissecting Nelson's XVR1. Nowhere have I said such. I started the thread to talk about filters, in particular filters that are easy to adjust. The simplicity of an S-K filter allows easy adjustment of frequency and Q. Since people seem interested in the opamp design Nelson is using, I posted a speculative schematic, although any opamp, discrete or chip, will do the job as long as it has high Zin, low Zout, and perhaps a bit of gain. Fred, as is his wont, feels that he has to 'save' people from me. I've known religious zealots who had better control of themselves than Fred does. (Incidentally, that, coupled with his prior tendency to strike from concealment using multiple [and false] usernames led to my infamous 'terrorist' comment. I still believe the comparison to be apt.) At any rate, I tried to keep the circuit simple, which is pretty much the philosophy behind Nelson's Zen series. Given that there are several posts that seem to be complaining that the circuit isn't complicated enough, I gather that I succeeded. Excellent! Filters need not be complicated. Use a buffer in front, the filter in the middle, and a buffer at the output, and you're done. The buffers can be more opamps, or even just followers, if you like. It's that simple. Anyone who tries to mystify it above and beyond that has an agenda, and it ain't Truth.
Can you design more complicated filters? Of course. Can you design other types of filters? Of course. This is meant to be a simple, adjustable, flexible filter. Nothing more, nothing less. Fred's insistance on trying to make it something that it isn't is based entirely on his emotional needs, and is not compatible with the stated purpose of the thread. It verges on threadjacking.
macka,
I used followers in my earlier thread on Sallen-Key crossovers a year or two ago. That's probably about as simple as you can get. Actually, I was using resistors instead of current sources underneath--even simpler still.
till,
I'm using the Motorola/OnSemi 2N5457 JFETs for my front end. Are they ideal? No. The 2SK389 is a better choice, overall. The last time I looked, I don't recall OnSemi having dual JFETs. Fairchild, I'm not sure about, although I seem to recall dual bipolars, perhaps. I did run across dual JFETs in the Mouser catalog; Siliconix 2N5564/5565/5566. I haven't used them so I have no opinion. They're pretty pricey. Given that my system is quad-amped, by the time you start adding buffers, low passes, high passes, etc. a $5-8 part gets pretty scary, although I might be persuaded to pursue it if they're good enough.
To folks here in the US, Happy Thanksgiving. To those in other parts of the world, eat well tomorrow and damn the calories. Enjoy life.

Grey
 
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