Active Crossovers (Analog and Digital) and Bi-Amping...

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Chops

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This isn't my first time doing this. In past systems, I have used active crossovers and multiple amps to power speakers before, both name brand speakers, modified name brand speakers, and DIY designed and home built speakers. So as far as active x-overs and multi-amp setups are concerned, I'm no stranger.

In those past systems, I used a dbx Project 1 223, Behringer CX3400, Rane AC22B, and finally, a Behringer DCX2496.

My thing is, and the reason I'm starting this thread is because that was 10 years ago when I last used active x-overs and such (Behringer DCX2496). The big thing these days are active crossovers in the way of DSP, now typically called "Speaker Management System" or something like that.

I know and understand the benefits of DSP, choosing any frequency and roll-off slope you want, better control and flexibility of everything, but the thing that concerns me is the AD/DA conversion. It's just a bunch more converting of the signal that I don't particularly care for. That's why the more I think about it, the more I just want to go with a good ole' fashion analog active x-over. And yes, I know the DCX2496 was this way.

As some of you may know, I have a pair of Martin Logan SL3's. I just recently starting vertically bi-amping them with a pair of updated/upgraded Carver M-500t MkII's, one of which I just received this past week. This alone has made a drastic improvement in stereo imaging, stage width and depth. Bass is also better. I was quite shocked at these improvements to be honest. So obviously there are benefits in bi-amping alone.

What I want to do with these ML's is completely bypass the internal passive x-over for the woofer since I have read repeatedly that this is the biggest and best upgrade you can do to these speakers do to the massive inductor utilized in the stock network. That makes perfect sense and I'm fine with that, but then there's the subject of the ESL panel...

Bypassing the passive network of the ESL panel is a bit more involving though also beneficial. For one, unlike the woofer, you can not simply remove the network and be done with it. The panel requires a power supply to operate it. It also requires not only a notch filter to help with its own resonance, but an EQ filter as well to help with the dipole roll-off. So instead of simply running a (+) and (-) wire directly to it from the binding post like you do with the woofer, you have to bypass the passive network but keep the power supply there, which has to have the amplifier signal run through it before it gets to the ESL panel. Makes sense, but a bit more complex. Then the issue of the notch and EQ filters.

Either I can leave the panel's network in place and just upgrade the capacitors in it for some improvements and leave it at that, or I can bypass it and do the above via an analog crossover and EQ, or get something like the DCX2496 or dbx Driverack PA2 or dbx Driverack VENU360... a.k.a. - a LOT of $$$ for that last one. Or I could go with the miniDSP and matching mic. But then I hear issues about the miniDSP being noisy or its signal being clipped or not having enough output, though it is very affordable and very flexible with the optional add-on apps that can be purchased and downloaded for it.

I'm not too concerned about room corrections either as I think the ML's sound pretty good as is in this particular room. Of course, there's always room for improvement, but to do room corrections, that means even more DSP processing and AD/DA converting, which I really don't want.

Lastly, IF I decide to keep the passive network on the panel, it's crossed over at 250 Hz at I think 12dB per octave. When I run the active x-over to the amps, what would I set the high-pass to, 250 Hz as well, or lower? If I set it to 250Hz, with an analog active x-over, that would make it a 48 dB slope. I'm guessing I would just run the woofer at 250 Hz, 48 dB as well, but not doable with an analog x-over.

Your thoughts?

Oh, and if you're running an active bi-amped system, could you please mention what x-over you're running and why, and maybe a brief run-through of the settings?

Many thanks in advance!
 
From another perspective...

I've gone through the process and ended up with passive bi-amping. Here's my thought process...
First and foremost, I'd like to keep things as simple as I can in the signal path. Active crossover introduces another active component in my signal chain, be it a DSP or analogue based crossover. Going down this path, the audio signal will go through lots of transformation before it reaches the amplification stage. Also, to reap the full benefit of the active route, I will also have to remove the internal crossover, which I am not ready to tackle.
My conclusion was to go the passive bi-amping route, and spend money on tube amps for the panels, and a really good solid-state amp for the bass.
Now, if I were to do maintenance on the ReQuests boards and I gather enough courage to remove the crossover circuitry, I'd go with the XM46 Passive Line Level Crossover from Marchand Electronics. Again, keeping things as simple as I can on the audio signal path.
I understand that the DSP-based crossover offers lots of options, but I hesitate when I know the audio signal will have to go through A/D conversion, digital processing, and D/A conversion to analogue domain. Too many stages which can rob the life out of the original audio signal.

Good luck with your system
Spike
 
Chops, don't be concerned about the DCX2496. It will not change or degrade the sound of your analog sources whatsoever. Here is the spectrum of it that came with my speakers.

Behringer spectra.jpg

It is performing with essentially zero distortion. All the harmonics (except the benign 2nd harmonic) are buried way down in the noise floor which is astonishingly low at 120 dB below the signal!

This is performance that even the best analog preamps would be envious of.
 
I've gone through the process and ended up with passive bi-amping. Here's my thought process...
First and foremost, I'd like to keep things as simple as I can in the signal path. Active crossover introduces another active component in my signal chain, be it a DSP or analogue based crossover. Going down this path, the audio signal will go through lots of transformation before it reaches the amplification stage. Also, to reap the full benefit of the active route, I will also have to remove the internal crossover, which I am not ready to tackle.
My conclusion was to go the passive bi-amping route, and spend money on tube amps for the panels, and a really good solid-state amp for the bass.
Now, if I were to do maintenance on the ReQuests boards and I gather enough courage to remove the crossover circuitry, I'd go with the XM46 Passive Line Level Crossover from Marchand Electronics. Again, keeping things as simple as I can on the audio signal path.
I understand that the DSP-based crossover offers lots of options, but I hesitate when I know the audio signal will have to go through A/D conversion, digital processing, and D/A conversion to analogue domain. Too many stages which can rob the life out of the original audio signal.

Good luck with your system
Spike

Hi Spike,

See, that's the thing. I like the idea of K.I.S.S. (Keep it simple stupid)... But, I know the benefits of the amps being directly connected to the drivers for best power transfer and driver control? And I believe those benefits probably outweigh any downfalls that may come with more gear in the signal path by a good margin.

So, it's either settling on the good improvements of simply bi-amping the speakers as is, OR reap the benefits of even greater improvements of going active.
 
Chops, don't be concerned about the DCX2496. It will not change or degrade the sound of your analog sources whatsoever. Here is the spectrum of it that came with my speakers.

It is performing with essentially zero distortion. All the harmonics (except the benign 2nd harmonic) are buried way down in the noise floor which is astonishingly low at 120 dB below the signal!

This is performance that even the best analog preamps would be envious of.

Hello Pneumonic,

It's not really a concern of degrading the sound of my analog as much as it is a matter of everything going through more converting. It's just a matter of the analog going through AD/DA converting in the first place, and the digital sources going through extra AD/DA converting.
 
Hello Pneumonic,

It's not really a concern of degrading the sound of my analog as much as it is a matter of everything going through more converting. It's just a matter of the analog going through AD/DA converting in the first place, and the digital sources going through extra AD/DA converting.
My point is that objectively the 2496 is benign sounding and won't add any audible sound degradation. So no need to worry.

BTW, going DXO was probably the best upgrade I have ever made in this hobby.
 
What's everyone's thoughts of the dbx Driverack Venu360? Reason I ask is because I just ordered one. From what I've read, it's supposed to be a big step up in sound quality from the PA series which I was thinking about getting, the PA2.
 
Well, got the dbx Venu360 in yesterday, and the matching dbx measurement mic today. Waiting on the XLR adapters, which won't be here until Monday.

Thought there'd be more talk on this subject in this thread, but guess not. Doesn't seem to be much activity on the forum at all lately.
 
What's everyone's thoughts of the dbx Driverack Venu360? Reason I ask is because I just ordered one. From what I've read, it's supposed to be a big step up in sound quality from the PA series which I was thinking about getting, the PA2.

Having used several active crossovers (Ashly, DCX2496, DBX DR 260, DBX DR 4800), I believe the Venu360 to be your best choice, as it has the same (possibly even improved if that's possible) 24/96 A/D/A that is also in my DR 4800, which is perfectly transparent. They are built like tanks, and run 24/7/365 for years on end with no reboots required. Quality costs money, and in the case of the DriveRacks, you definitely get what you pay for.

I see you bought the DBX mic, however, the internal measuring scheme is designed for auto-adjsutments at venues, not fine tuning in a home setting. I'd rather use a REW setup on a laptop and a Umic to do the measurements, then manually set the EQ.
Auto-EQ systems are usually confused by a dipole electrostat. They miss-read the reflected energy and overcompensate. Even if manually measuring, that's something to be aware of as well.
 
...
What I want to do with these ML's is completely bypass the internal passive x-over for the woofer since I have read repeatedly that this is the biggest and best upgrade you can do to these speakers do to the massive inductor utilized in the stock network. That makes perfect sense and I'm fine with that, but then there's the subject of the ESL panel...

Bypassing the passive network of the ESL panel is a bit more involving though also beneficial. For one, unlike the woofer, you can not simply remove the network and be done with it. The panel requires a power supply to operate it. It also requires not only a notch filter to help with its own resonance, but an EQ filter as well to help with the dipole roll-off. So instead of simply running a (+) and (-) wire directly to it from the binding post like you do with the woofer, you have to bypass the passive network but keep the power supply there, which has to have the amplifier signal run through it before it gets to the ESL panel. Makes sense, but a bit more complex. Then the issue of the notch and EQ filters.
Yes, the ESL side is more complex, but not really that bad, it's more about not cutting out the feed to the energizer board and ensuring the right polarities are preserved feeding the step-up transformer. And ensuring a sufficient impedance load is present between step-up xformer and amp, as some amps just don't like the low-impeadance, reactive load. So a 1-Ohm (60W) load on the positive feed to the xformer is recommended.

As for the notch and EQ, don't worry, just EQ to the actual post-passive setup. If a notch is required, you'll see exactly where and how wide on the measurements using REW. Same for EQ. But be careful, the ESL HF has a significant drop-off, don't try and over-compensate for the natural decrease above 12khz. EQ can help in the lower mid-range to compensate for some of the dipole cancelation, but moderation is again the watchword.

If you measure and EQ each channel with the speaker in its intended listening location, then you also benefit from a degree of 'room correction' as the EQ's could be quite different depending on positioning.


Sounds like a fun project, do keep us posted.
 
Having used several active crossovers (Ashly, DCX2496, DBX DR 260, DBX DR 4800), I believe the Venu360 to be your best choice, as it has the same (possibly even improved if that's possible) 24/96 A/D/A that is also in my DR 4800, which is perfectly transparent. They are built like tanks, and run 24/7/365 for years on end with no reboots required. Quality costs money, and in the case of the DriveRacks, you definitely get what you pay for.

I see you bought the DBX mic, however, the internal measuring scheme is designed for auto-adjsutments at venues, not fine tuning in a home setting. I'd rather use a REW setup on a laptop and a Umic to do the measurements, then manually set the EQ.
Auto-EQ systems are usually confused by a dipole electrostat. They miss-read the reflected energy and overcompensate. Even if manually measuring, that's something to be aware of as well.

Well maybe I can use the dbx mic for the subwoofer and woofer sections. If that proves to not be beneficial, then I'll resell or trade it for a USB mic.
 
You really need to use a separate measuring rig like REW to be able to analyse the measurements of each portion (panel, woofer, sub) and chose appropriate crossover points and slopes, as well as any EQ a the driver level. Then you measure the entire system in-situ and devise additional EQ correction to get rid of the worst room-induced effects (typically under 500Hz).

Being able to see not just frequency response, but critical info like the Impulse response are necessary here. The Impulse responses of the panel and woofer need to be aligned both in time and in phase.
Waterfall plots let you analyse excess-energy problems in panel mounts, woofer enclosure and ultimately, the room.

A USB mic is the simplest way to use REW, but if you a phantom-powered mic like the DBX RTA, you can also use a USB IO box like the Tascam US-366 I use for my calibrated mics. But a Umic is cheaper.
 
Yeah, I'll just go with a USB mic. It's simpler and definitely cheaper.

It's been a while since messing with a DSP crossover, so I have some reading to do as well as reading on the procedures you mentioned above. No worries though. It's only a matter of time. And speaking of time...

I just received the XLR/RCA adapters in the mail today. So tonight, I would like to integrate the Venu360 into the system. I won't be able to open up the SL3's until this weekend, but in the meantime, I'd like to try out the Venu360 and get accustomed to it via it physical controls as well as the ones via the laptop and/or iPad. Since the passive crossover points in the SL3's are at 250 Hz, I'm going to set the Venu360 at 250 Hz as well. The question is, should I have it at 1st order, or go higher? Either way, it's going to be making the crossover point steeper than the stock 2nd order.
 
Okay, so some exciting times tonight.

I installed the Venu360, ran the cabling to the amps, adjusted the x-over, turned the amps on, slowly turned up the preamp's volume, and sound emitted from the SL3's!

I set the crossover points to 250 Hz, LR 4th order for the woofers and panels (defaults were a combination of BW 2nd order and LR 4th order). The SL3's sound better all set to LR 4th order. I have the high-pass of the woofers set to 16 Hz. If I run them flat out, they end up cancelling out with the sub, requiring them to be inverted, in which case makes the bass a bit too bloated.

I have the JL sub set to 45 Hz LR 2nd order.

Something I've noticed is a tiny tiny bit of hum when up close to the sub and woofers of the SL3's. With the Venu360's inputs set to the default +24 dB, it's barely noticeable, but when I set the input to +14 dB (where it probably should be at), the hum is very noticeable, even at the listening seat. So I have to figure that out.

Of course, these are just quick preliminary settings. I won't get truly serious with it until I open up the SL3's and bypass the passive networks. Until then (probably till this weekend), these settings will do.

On a side note, my inner youth has been getting the best of me, listening to albums like "The Cure: Mixed Up" and "Depeche Mode: Violator" and turning on the Subharmonic Synth. Takes me back to my car audio days and using the Audio Control Epicenter! :rocker:

Anyway, a few pics...
p2336151225-5.jpg


p2336151226-5.jpg


p2336151227-5.jpg


p2336151229-5.jpg
 
Here's a couple pics of the guts of my SL3's. Well, one of them anyway. Just not feeling up to it today, so no motivation to do anything. I wasn't feeling too good yesterday either... Plus I need to get some supplies before I can do any work on them.

p2338925021-5.jpg


p2338925006-5.jpg


p2338925023-5.jpg




And just for fun from last night...

p2338924775-5.jpg


p2338924779-5.jpg
 
Okay, I decided to go ahead and bite the bullet. Instead of only bypassing the woofer crossover, I went for the gold and bypassed both crossovers. I'm currently using crimp-on butt connectors as a temporary thing, just for proof of concept so to speak. I didn't want to take the time of soldering and such in case something didn't work out right. Plus the fact that my current soldering iron is kaput anyway. I'm going to do some shopping around and buy a nice Weller soldering station soon.

I've only done the left channel so far so I can compare to the stock right channel. I'm using a Diana Krall album that I'm extremely familiar with as the reference. I can definitely tell there's going to be some needed tweaking done, especially on the ESL itself, but nothing too drastic. The biggest difference is that the entire left channel is probably a solid 3 dB louder, which goes without saying, mainly in the panel section. And maybe this is due to the extra loudness, but the left channel seems to be a bit more lively and snappy, or in other words, more dynamic.

I guess it's now time to power down the right amp and pull that SL3 apart. In the meantime, here's some pics of the process. It was a LOT easier than I had originally thought!

p2344636723-5.jpg


p2344636724-5.jpg


p2344636727-5.jpg


p2344636400-5.jpg


p2344637099-5.jpg
 
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And just as simple as that, the second SL3 is done. The first one took me nearly 45 minutes to get through and do. This one... 10 minutes. LOL

p2344746429-5.jpg
 
My Martin Logan SL3's have dynamics! Something they never had before! In fact, that was the only thing I didn't like about them, was the lack of dynamics.

So far, the dbx is just running crossover duty only. The panels are crossed over at 250 Hz with a Linkwitz 48 dB slope, the woofers are also at 250 Hz, but with a Linkwitz 12 dB slope and 16 Hz on the bottom end with a Linkwitz 48 dB slope. The sub is set to 46 Hz with a Linkwitz 24 dB slope.

I'm not quite sure why, but when I initially had the SL3 woofers crossed over at 48 dB @ 250 Hz, there was a big hump in the mid-bass, making everything rather bloated and chesty sounding. Knocking it down to a 12 dB slope eliminated that hump and made everything sound natural. There's more body to vocals like there should be (which is something the SL3's are known to lack a little of).

Bass content now has snap and texture. Vocals, both male and female sound amazingly real and lifelike. They have life and warmth to them. Treble is smooth and detailed. With no EQ of any kind, they sound like they're being powered by a good tube amp without being rolled off.

The JL sub has completely vanished into thin air. It has become a natural extension of the SL3's. With your eyes closed, you just assume all of the bass you're hearing is coming from the SL3's and nothing more. Then again, with your eyes closed, you really can't tell that any of the sound is coming from the speakers. It's just there in the room with you and that really low bass just shows up when needed.

Listening to Louis Hayes' Serenade for Horace right now, and it's oh so good sounding!
 
I knew you would like the active bi-amp setup. Now that you've bypassed the panel crossover, you should check the panel/woofer phasing. I don't know an easy way to do this-- but you could play a 250 Hz test tone with the woofer leads inverted and not inverted. When phasing is inverted the respective panel/woofer wavelengths blend destructively and the output diminishes. Conversely in phase wavelengths blend constructively and the sound output increases. Whichever way plays loudest is correct.
 
I knew you would like the active bi-amp setup. Now that you've bypassed the panel crossover, you should check the panel/woofer phasing. I don't know an easy way to do this-- but you could play a 250 Hz test tone with the woofer leads inverted and not inverted. When phasing is inverted the respective panel/woofer wavelengths blend destructively and the output diminishes. Conversely in phase wavelengths blend constructively and the sound output increases. Whichever way plays loudest is correct.

Hi Charlie!

I knew I'd like the active bi-amp route as well. Been there before, just not with these speakers. The speakers I had done this with before were a DIY pair designed by yours truly. I had only ever heard them active.

These SL3's on the other hand, I've been listening to them for over a year bone stock. So to hear what they are really capable of when "setting them free" via active bi-amping, it's a real eye opener. And there were no drawbacks at all going active either. Yes, there's more money involved, but it's definitely worth it. It's just a major speaker upgrade for a fraction of the price of an actual speaker upgrade.

As for the phasing, I've done lots and lots of listening of familiar recordings, and have been flipping the phase back and forth numerous times via the Venu360 PC/iPad/Tablet apps. I'm 99.9% certain I have them properly phased. Of course, once I get a USB mic (have a miniDSP mic being sent to me), I'll fire up the REW software and make sure.
 

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