How are active crossovers so simple/clean?

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Hi all,

This is a newbie question. If it's a FAQ, please can you just point me to where I can get my answers?

My question is about the apparent simplicity of active crossovers versus traditional output-level (as against line-level) passive crossovers. (I'll call them POL XO for passive o/p-level...)

When we see POL XOs for "real" speakers, we often see a lot of complexity and asymmetry at the XO point. For instance, in a woofer-mid XO, you may find that the high-pass for the mid is an electrical second order while the low-pass for the woofer is a fourth-order. This is because the electrical 2nd-order high-pass for the mid is working in conjunction with its sealed enclosure, resulting in an accoustic 4th order. And so on.

With some other speakers (very good ones too), the XO has been designed such that the high-pass and the low-pass don't even cross over at the same frequency... the gap is wider than what the pure electronics theory would necessitate. However, accoustically, there's no dip in the frequency response graph of the speaker, because the humps and peaks of the drivers' response curves are suitably compensated by the wider-than-natural gap of the electronics.

In other words, the electrical XO circuits are rarely simple and symmetrical, yet, in finely tuned designs, the accoustic results are smooth, uniform, and give you a beautiful flat response curve.

However, when I see (analog) active XO based designs, I find people plonking in a 2nd or 4th order LR XO, same slope on both sides of the XO point, no strange wide gaps between the two curves, etc. And they claim it sounds lovely.

My question is: how come we don't see similarly complex, asymmetrical active XO designs? Is it that all the active XO designs I've read about have less-than-mature XOs designed? Is it that no one has designed finely tuned active XO based speakers of the level of complexity and fine tuning as, say, the Ariels? Is it that active XOs make many problems just disappear, simply because there's no POL XO in the path? I suspect it's a bit of all of the above, and some more...

Please can you help? I was thinking of embarking on the design of a new speaker (open baffle to boot)...

Tarun
 
Is it that all the active XO designs I've read about have less-than-mature XOs designed?

Not all, but many. Maybe most. I've seen some published designs with proper eq, signal delays, and such in Speaker Builder. And of course there's the Linkwitz designs.

Part of it is that one doesn't have to worry as much about the reactive charactristics of drivers. That factor as much as driver acoustic response is responsible for the asymmetry of many passive designs.
 
SY said:
Thanks for pointing that out. It's now in its proper place.
Thanks, SY.

Going by your answers, does this mean that if I design, say, a three-way speaker (without any XO, but with separate leads for each driver at first), and then test the frequency response curve for each driver, and verify that there's there's a 2-octave overlap between the flat portions of every pair of drivers (i.e. the woofer to mid, the mid to tweeter), I can simply put in steep 2nd order or 4th order LR filters and get a flat response curve from the speaker? (I'll do SPL matching later on.)

My reasoning is that if I operate well within the measured flat areas of the drivers, I'm taking into account all the driver+enclosure interactions in my measurements itself. And crossing over at points where both sides are flat, I don't have to bother about compensating for various types of slopes of the various complex driver+enclosure interactions. Does this sound ... well, sound? :)

Tarun
 
Tarun, correct xover design is a very complex issue, particularly for a passive network.
There is no such thing as a set of drivers in a box that have a substantially flat response, particularly for 2 octaves either side of the xover point. They always need eq as well as filtering. Even if these existed, the phase shift associated with a roll-off 2 octaves away will still affect how well they sum togehter.
A passive network has the extra complexity of working into a highly reactive load (the driver) and so needs extra components to help deal with this, and must also present a reasonable load to the amplifier, without the impedance dropping too low or being too reactive.
Using simple active filters with no active eq simply shows a fundamental lack of knowledge of speaker design. It has been shown numerous times in the literature that the target function for the speaker drivers to sum correctly acoustically is that the combination of driver AND xover should be , for example, 4th order Linkwitz-Riley (if that is your preferred choice). Even this will not sum correctly unless you take into account driver delay due to positioning on the baffle.
The desired acoustic response, and hence the required electrical response, is independant of the requirements of matching the passive xover to the reactive load of the driver.
Hope this helps clarify your confusion
 
Yeah, what Andrew said ;)

In addition, you'll need to think about the interactions of the polar response of the drivers and crossovers. There are various schools of thought on what polar patterns ought to look like and how they should vary with frequency; before starting the crossover design, you need to decide that fundamental issue as well.
 
Whew!

AndrewJ said:
Tarun, correct xover design is a very complex issue, particularly for a passive network.
There is no such thing as a set of drivers in a box that have a substantially flat response, particularly for 2 octaves either side of the xover point.

Wow. Thanks guys. You blew all my optimism and most of my confusion away. :)

But let me try to pick up the pieces, just for my own sake. First, about the "drivers in boxes with substantially flat responses" bit.

Isn't it possible for me to choose, say, a woofer driver and put it in a box, such that its response will be substantially flat from, say, 60Hz to 400Hz? I was under the impression that it would be possible with some patience but without Linkwitz-like or Olson-like expertise. Similarly, I'll carefully select a midrange and either mount it in an OB or put it in a sealed box such that the midrange + box will give me a largely flat response from, say, 200Hz to 3KHz, when measured. Just by itself. No XO, no integration with other drivers, etc. Is this possible?

I am assuming, of course, that driver selection will be done carefully, and the box will be designed using modelling tools and other software, and may even be fine-tuned iteratively to achieve a driver+box combo which is flat in a range.

This was my first premise. Please guys, can you react to this? I don't see any point in my going ahead with this line of thinking if my most basic premise itself is overly simplistic.

If it works this far, then in my next post I'll ask about XO. And AndrewJ, I'm never thinking POL XO. I'm only thinking of active XO, because that's the only thing I can remotely understand.

Thanks enormously,
Tarun
 
Two problems: the woofer is in a room, and the midrange is on a baffle. The responses and polar patterns will vary accordingly. Now, all of this can be modelled, and current software does a fairly good job of it. You can use this to get an idea of what effects your design choices have on driver response.

But at the end of the day, what you need to do is get your drivers in boxes and on baffles and measure what you've got. Then the electronic crossover can be designed properly with the realities of your implementation in mind.

No, doing a good design isn't easy, and a 3-way is MUCH harder to do than a 2-way. But it IS possible, it's a fun challenge, and you'll learn a lot. Don't be afraid to make some mistakes on the way.
 
SY said:
But at the end of the day, what you need to do is get your drivers in boxes and on baffles and measure what you've got. Then the electronic crossover can be designed properly with the realities of your implementation in mind.
Okay... If I'm reading you right, it means that my pre-building analysis and modelling may not be of much use in predicting the in-room response curves. Fine. In that case, let's assume that I'll do "the best I can" with the modelling etc, and then build and measure in-room responses. (I'd intended my measurements to be in-room anyway.)

Once this is done, is it really possible to just build eq circuits to smoothen out the dips and peaks of each driver? What about phase response non-linearities introduced by the eq? Can they be ignored or are these distortions very audible? Also, please note that I've not even begun talking about the XO... I'm still trying to get individual drivers to deliver flat frequency responses. :)

Is this how you would go about doing a scratch design of a speaker? Or am I fundamentally wrong about my starting point itself?

Also, why is 3-way so much more difficult than 2-way? Does it still remain difficult if I use a so-called FR driver as my mid, and only use woofers for lower than, say, 200, and tweeters for higher than, say, 6KHz?

I keep coming back to the old line of thought, which I can't get out of my head: if I can get "substantially flat" stretches in the frequency response curves of individual driver+box, and their respective flat stretches overlap by 2 octaves or so, is it easy to build an active XO to put them together?

Sorry about being so bull-headed persistent, but I really want to know... :confused:

Tarun
 
You're getting the idea. Let me walk you through the way I designed my last set of dynamic speakers, which were quad-amped four-ways.

First, I chose the basic configuration and design goals. In my case, it was seperate subwoofer modules and three-way "satellites." I wanted the satellites to be free-standing and have their woofers floor-loaded. I wanted the vertical dispersion in the mids and treble to be relatively narrow. I wanted horizontal dispersion to be fairly flat out to 30 degrees, then roll off smoothly in the treble thereafter.

I chose my drivers from some experience working with the units. I chose to use a narrow baffle just to simplify that aspect and for esthetics. If I were to do this one again, I'd play around with BDS models to see how good a choice that was. The vertical dispersion was handled by using an MTM.

Next, I drew up the cabinet plans and farmed out the wood-cutting. Once the wood was back, I mounted up the drivers and ran leads out. I hooked up my test setup (FFT) and took LOTS of frequency response curves, on and off axis, for each of the drivers (I ran the midranges in parallel). Nearfield below about 250 Hz, 2m above. Measurements were also taken out in a large parking lot, just to verify that my gating wasn't introducing anything odd.

Now comes the "art," which is choosing the crossover points, slopes, and designing the EQ. My choice was to go with what I know, or to examine similar commercial designs and see how they did things. I did the latter, then went back to the former. The trade-offs are power handling and dispersion. I went with 4th LR (acoustic slopes!) all the way in this case.

Then build. Then measure again. Then tweak the amp gains and crossover values to hit the design target. Then adjust amp gains to get relatively flat in-room power response, using a third-octave spectrum analyzer. Then tweak by ear, which took for-EVER.

The easiest part came last. I enjoyed them for about a month. Then someone offered to buy them, which helped me fund my current ESL project.

And that's all there is to it. About 6-7 months of evenings and weekends compressed into a few paragraphs. A nice two-way would have taken about 4 weeks max.

If you don't have them already, do procure copies of d'Appolito's book on measurement and Dickason's LDC. You'll need a good test mike (mine was cheap, based on the Panasonic electret capsule), an FFT/MLS analyzer (and know how to use it), and a real-time spectrum analyzer (I used a software/soundcard version).
 
My prejudiced, unscientific, personal opinon

is to suggest that you do what I did with my active subs.

Buy a Behringer DCX 2496 - $325 us to your door if you're in the US.

Essentially use it as a digital crossover - you can do digital in if you want.

You've got the flexibiility to do what you want -

Bessel, Butturworth, L-R up to 4th order slopes, delay, auto align, choose crossover points, slopes at will.

Invert phase, set phase at any compass point, etc.


You have to use XLR connectors.

I was hesitant to buy the Behringer due to a poor reputation on analog equipment.

However, the digital stuff seems OK. Thorsten Loesch is a fan of their Ultracurve.

You also have the ability to EQ each driver -

I was surprised to see how much difference 2 points of less than 2 db EQ did.

When you're getting close, 10 hz can mean a noticeable difference.

Caveat: I am still using a 10 or 12 year old reciever for a preamp - there could be some noise or artifacts that it is not reproducing.

however, my horns and drivers are a little over 100DB in efficeincy, and when you get that high in sensitivity you usually hear any problems in the chain ahead of you.

I can hear no problems with the 2496 in the system - quite the opposite, it sounds much better in the system.

Since you're talking active - you might seriously want to consider it.

I would not recommend the unit for pro use - it seems too lightly built - however, it seems fine for a set and forget crossover at home.

YMMV -

One poster in a pro forum said he had heard that the units were noisy - however, none in mine - And none in the users that I know of that are using them at home.

Do a search here and at AA for DCX 2496 behringer

Regards

Ken L
 
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Hi Tarun,

I'm not sure how much of a newbe you are, but I would take this attitude. I'm a newbe, so I know what I make won't be perfect, but will probably be a lot better than if I spent the same amount of money on a comercial speaker. I'll learn a lot and it should be a lot of fun (even if at times a bit frustrating)..

The speakers I currently have look terrible, have some major design flaws, and I'm sure most of the experienced guys here would cringe if they heard them. That's why I want to make some new ones. However most Un-enlightened people who hear them think they sound wonderful, compared to the usual dept store stereo gear that most people buy. I also have a freind who has spent quite a bit of money on HT gear who recently heard them and was amazed.

I know they are pretty ordinary, but to the average person they are great. As I saw someone else say, If you design the perfect speaker first go, then you do yourself out of a hobby :).

Don't fret. Design build enjoy and learn :)

Yeah I know I should take my own advice :)
Regards,

Tony.
 
Thanks

SY said:
You're getting the idea. Let me walk you through the way I designed my last set of dynamic speakers, which were quad-amped four-ways.
Thanks very much. You've helped me clarify my line of thinking now.

I have the following questions. Many of them are to do with your choices rather than my conceptual confusion; I found your description very interesting. Made me think quite a bit.
  1. What software do you use for modelling speakers, getting box behaviour, and other things like BDS? I know the usual links and lists of packages, but I wanted to know your personal recommendations.
  2. Does one need FFT software and spectrum analysers? Can't one do the same thing using tone generator CD and SPL meter? (I made myself a tone generator CD with six tones per octave, and I have the Radio Shack SPL meter with a linearising table I downloaded from the Net somewhere which specifies the error in measurement at each frequency.)
  3. How did you check for phase linearity among drivers and phase distortions due to your XO?
  4. While looking at "similar commercial designs", did you get to study active speakers or did you have to map from passive designs? How easy or difficult is it to move from passive to active XO?
  5. What XO points did you decide on? Will you tell me a bit about your experiences with shifting of XO points during fine-tuning?
  6. Why did you do your measuring in an (admittedly imperfect) anechoic environment? Why not in-room? Was it just to feel better that your speakers had been tuned flat in a "neutral" environment, in case you needed to move house (or sell the system, which you did)?
  7. How complex was the XO? Did you actually have to do things like build an electronic second-order Bessel to get an accoustic 4th order LR, etc? Is there any modelling software which tells you what electronic XO will give you 4rth order LR? Approx how many opamps did your design eat up? Did you make PCBs for your XO and eq circuits?
  8. Did you face a lot of problems with interactions between the high and low XOs? You keep mentioning that a 2-way would've taken much less time...
  9. Would it have been easier to build a first-cut set of XO based on electronic-to-accoustic modelling, and the just apply something like the Behringer Ultracurve to take care of all the smoothening eq? Would it have given results as good as what you got? Do you think SL's Phoenix or Orion design process could have been simplified and given as clean a result with an Ultracurve or a dbx Driverack? The more I think of active XO+eq, the more I feel that opamp-based analog circuits are a pre-Industrial-revolution way to do things. Analog circuits for XO might still be okay, but what abt all those peaks/dips to be eq'd out? And then comes in-room eq... What do you feel?
  10. Suppose you built your first-cut active XO circuit using analog active filters, and also suppose you had a 10-band parametric equaliser. Would you have put this before the active XO and done all your eq'ing with it? Would it have been as good as what a custom-built SL-style eq circuit achieves?
  11. Finally... how good did it sound? What did you feel would have been done differently if you had more time/experience/money?
Re-reading the list of questions, I now realise that asking you to answer them is asking for a lot of patience. I'll be happy with whatever you can spare.

I'll get both the books you suggest... I have neither. I've heard about both.

Thanks once again. Your two or three posts have been more useful than a book for me.

Tarun
 
Re: My prejudiced, unscientific, personal opinion

Ken L said:
Buy a Behringer DCX 2496
[/quote]
Amazing. Ken, you beat me to this post by three hours. :)

Quite amazingly, I had been planning to respond to SY's deeply insightful post, and had the page open on my screen right through the night. Got up this morning, and directly started typing my response. I didn't refresh the page, so I didn't see that your post and one more had come in during the night. And you beat me to the digital XO idea. :)

Thanks for your post. I was beginning to veer strongly to your point of view; you've strengthened it now. I'll wait and see what others have to say. In fact, I'll wonder what SL himself would have done with the Phoenix or Orion today, with the Driveracks and the DCX 2496 available. Would he have made PCBs, or would he just have given you a datafile to upload to your Driverack? Change one file and you go from 6" side wings to 12". Change to a second file and you replace those Peerless drivers with Adire Tumult. Change to a third file and you go from Phoenix to Orion! Magic! :D

In fact, if you see my post just before this one, you'll almost be able to see my brain ticking... :)

Tarun
 
tcpip, I'll have to answer this in pieces over the course of the evening, since I'm watching our 2 year old while Mama is working.

First, I absolutely concur with Ken; the next dynamic speaker I design, I'll use the Behringer unit. It sure beats all the soldering and unsoldering I went through!

Second, to your specific questions:

1. I use some obsolete stuff because that's what I've got and that's what I'm familiar with. IMP and Calsod are the backbones, and in the past year, I've started using BDS and polar pattern software from the FRD Consortium.

2. Yes, absolutely, to do it right you need something like FFT/MLS. But with modern soundcards, that's not too big of a hurdle. Using it properly is the key, and the d'Appolito book will take you far.

3. Drivers and filters are generally minimum phase, so there's no phase "distortion" per se. Acoustic center offsets are important to know and trivial to determine with the FFT systems. Other phase deviations from minimum phase are normally due to reflections and diffraction, which is easy to spot with FFT units.

More in a little while. Thanks for your kind words.
 
wintermute said:
I'm not sure how much of a newbe you are, but I would take this attitude...
Thanks. I know exactly what you mean. And I can assure you I'm a total newbie. I've just been doing reading up, that's all. The first speakers I ever made were a mono boombox using a TDA2020 amp and a single dual-cone FR driver in a ready-made particle-board box which used to vibrate and rattle merrily. This was in 1985, when I was in the second year of my undergrad studies. The second speakers I made were small, simple, ducted port bookshelves copied from Jordan's Website, based on the JX92S; this was last year. In the process I learned how good the JX92S is. I also learned that most un-enlightened visitors to my home would prefer my Wharfedale floorstanders to the JX bookshelves, simply because the Wharfedales had a room-filling bass extension which probably no bookshelf can match. No one paid attention to the better clarity and detailing of the Jordans. That's when I realised that I need to be a bit careful with my R&D. :)

You're right about this being a hobby, and of course I agree that no amount of pre-reading will teach me much unless I try to experiment. But perhaps we Indians are a bit more careful before spending for parts, because our disposable incomes are much smaller than those in developed countries, if converted to a common currency? I don't know. I only know that my music system (which I paid about USD 2000 for, and which you can buy in the US for perhaps USD 1200) is more expensive than that of any friend or relative. In the US or Europe it would be "mid-fi." :)

But the learning is fun. Specially if we can tap the wisdom of those who are ahead of us, like you. :)

Tarun
 
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