Hi Everyone,

The LFR1100 Active is a very special loudspeaker system, but it's also much more complex from a component and wiring standpoint than a typical setup. I thought a little "best practices" guide might be helpful. I’m also happy to answer any of your LFR1100 Active questions in this thread.

Components

The LFR1100 Actives come with a separate DSP chassis for each channel. When selecting partnering amplification, it's best to dedicate a single 5-channel amplifier (or a combination of amplifiers to make up five channels) to ONE speaker channel. That means one DSP to one amplifier. Combining connections from two DSPs to a single amplifier can sometimes result in signal ground loops that cause buzzing or hum. Ideally you would also not share the LFR amplifiers with other speaker channels in your system for the same reasons (like using a 6-channel amplifier for one LFR and your centre channel).

Wiring

If you’re taking advantage of the custom-made, color-coded wiring loom that we’ve made available for the LFR1100 Active, it might be confusing to come up with the correct interconnect lengths without having the components in front of you. If you’re stacking your amplifiers on top of the DSPs, you would want an interconnect length of 18 inches. Likewise, if you’re going to have your amplifiers and DSPs side-by-side on a shelf, you want an interconnect length of 36 inches, plus whatever spacing you think you’ll have between the components. As for which type of interconnects to use, we strongly suggest using balanced XLR connections if both your pre-amp/processor/receiver and your amplifier(s) have them available. You should never mix XLR and single-ended RCA connections on the DSPs as you’ll experience a gain mismatch. All interconnect cables should be kept as far away from AC cables as possible.

Placement

Omnidirectional loudspeakers like to have some breathing space behind them to perform their best, but it’s not necessary to have them many feet out into the room. Obviously, placement is going to be highly dependant on your actual room, but the DSPs allow you to tailor the amount of rearward information via the Boundary Compensation switch. This control is also very useful if you have an asymmetrical room layout, or a rear wall with openings like doorways, etc., as it will allow you to balance the sound between the left and right channels. For the electronic components, it’s best to have each chassis on a dedicated shelf with adequate air space above the amplifiers. If you need to stack components, it’s best to have the DSP under the amplifier for proper ventilation. Cables should be routed in such a way to keep low-level interconnects together and away from AC cables. It’s also a good idea to have all of your components on the same AC circuit as it will minimize the chance of ground loops.

Thanks and happy listening!

Andrew

Thanks Andrew. Very useful.

I had been planning to use RCA from pre-amp to DSPs and XLR from DSPs to amps... how bad would this be ? I did have a vague nagging thought that XLR signal levels were traditionally higher than RCA levels but was thinking that just meant I shouldn't be using XLR-to-RCA cables anywhere.

Thanks Andrew, great info! I do have a ADA1500 channel assignment question for you.

If I had an ADA1500-8, there are 4 amp boards in it and each board is limited to the power it can put out. So, If I were to connect an ADA1500 to one, and also want to power 1 surround (say a QS10HP), and a rear (say a QS8), what would be the optimal connection strategy? I was thinkin the following:

Board 1 - LFR1100 Front Bass, Second Slot Unused
Board 2 - LFR1100 Front Mid, LFR1100 Rear High
Board 3 - LFR1100 Front High, LFR1100 Rear Mid
Board 4 - QS10HP, QS8

Thoughts?

With RCA in to the DSP and XLR out you will be fine, you'll just have 6dB of additional gain which will likely result in the same output level from your amplifier as if you had used the RCA input (assuming your amp has a truly balanced input). The other way round (XLR in and RCA out) would result in a loss of 6dB of gain and you would probably clip the DSP before being able to drive your amp to full power. Our DSPs are unity gain so that 1 Vrms in equals 1 Vrms out RCA to RCA or XLR to XLR.

Thanks,

Andrew

Having the bass on it's own amp module is a good idea, but I would recommend having the front high and mid and the rear high and mid together on one amp board, don't mix front and rear. It's cleaner from a path back to the star ground in the DSP that way and will likely result in a slightly lower noise floor.

Thanks,

Andrew

Super, thanks Andrew!

Thanks Andrew. The amps are ADA1500's so I imagine they should be OK.

I will be mixing channels since I originally bought an ADA1500-4 for passive LFRs... the plan is to use that for the woofers and I ordered an ADA1500-8 for the mids & highs.

Sounds like I should keep mid/high for each of front/rear/left/right on the same amp board (guessing that's two adjacent outputs) ?

Even more useful information, thanks !!

Hi Andrew, this is a pretty basic question. I’m sure I have missed it, but I’ve been trying to find the dimensions of the DSP unit. I will be stacking each with an ADA 1500 and I’m trying to decide on placement. What is the height dimension of the two stacked on top of one another? Thanks

Yes, with ADA1500s you will be fine. If you can avoid mixing channels from the left and right DSPs on the same amp module, that will be best. The two channels from one amp module go to each grouping of four binding posts.

Thanks,

Andrew

The total height of an ADA amp stacked on top of a DSP is approximately 9 3/8", including the feet.

Thanks,

Andrew

Perfect, thanks !

Thanks Andrew. This is helpful.

At the risk of asking a dumb question, for a stereo setup is there a benefit to running a 1500-4 with one module being used for each of left and right channels vs a 1500-2 which presumably has one module for both ?

I have a 1500-4 so I'm happy no matter what the answer is

There is no benefit if you are only using two of the four channels of the ADA1500-4. For the ultimate you would need to step up to the "Ian-approved extreme stack" of an ADA1500-1 and two ADA1500-2s for each LFR1100 active.

Thanks,

Andrew

Andrew I have a sub wiring question. I'm going to drop down to one EP800. I would prefer to use the output of the LFR1100 DSP and tell the Denon that there is no sub. I would set the Crossover of the EP800 at 40 Hz and then adjust the level as necessary. In order to get both the L and R sub output, I was thinking of taking the output of both the DSPs and putting them together via a Y splitter and then sending the combined signal to the EP800 input. Is there any reason not to do this?

SRoode, that won't work. Splitters can't be made into "adders" by reversing their connections. Circulating currents will be created between the DSPs.

I suggest you use the receiver's crossover. The receiver will also allow you to set distance which the DSP does not allow. I've raised this to Axiom but I don't know if it will ever be added. Without a distance setting, a sub fed off the DSP cannot be placed further from the MLP than the LFRs.

There's another thing about sub integration worth noting. I have two 500s fed off the DSPs crossed at 40Hz on the DSP. The actives are running full range at present. I have the 800 on the .1 channel. Up until yesterday, I was wondering what the phuk happened to my infrasonics. How could it be that I went from M100s to active LFRs and lost my infrasonics? Then I recalled how much wonderful bass I had in my living room and had a thought. I toed the LFRs in by 15 degrees and BAM! the infrasonics have been unbelievable. The LFRs and subs were destructively interfering with each other. This also explains why the 500s had no effect in my living room; the way they were positioned, relative to the LFRs, led to cancellations.

Thanks Mojo, that is excellent input! I always toe my mains a bit so that's not a problem.

My point is that sub integration with the actives may be challenging. Toe in helped me but it may not help you. Toe in helped with infrasonics but I am not happy with the soundstage for music. The battle continues. One day I may be as happy with the actives in my basement as I am with the M2OW in my living room. That day is far off.

When setting up subwoofers, particularly within the context of a stereo system, I have never concerned myself with group delay or the absolute difference in phase angle between the mains and the sub(s) due to differences in distance. Why? Because I have not found that it has made much of a difference. My basic setup technique is to get the main speakers placed where they are sounding good and place the subwoofer where it's convenient. Play a STEREO track with well recorded acoustic bass, preferably plucked like you will find in many jazz arrangements. Something with a good bass solo or minimal instrumentation is ideal. Turn it up to a decent but not loud level and go over and kneel by the sub. Set the phase control to the zero position and move your head toward the main speakers. Listen to the tone of the bass notes. Now move the phase control a quarter turn. Did the tone of the notes get more defined? Are they thinner or fuller? If they are thinner the original position was best. I find that in many setups I can actually rotate the phase control slowly and you will hear a point where the bass between the mains and the sub "locks". Now move to your main listening position and repeat the track. Does the bass sound full and well-defined? If it does you've found your sweet spot.

Thanks,

Andrew

Andrew, thanks for trying to help. I really wish it was that easy. Just stop and think about this: how can I verify that the tens of thousands of $ of gear I have in my basement works together as an optimized system? Think about all the physical and digital knobs and switches that exist within my system. How do I adjust them to optimize? How do I know that I don't need more knobs and switches with perhaps higher resolution? I don't expect an answer to this but I think it's worth thinking about on your end because it presents an opportunity.

I've read countless of times on this board comments like "I love my Axiom subs. They are totally silent until there's something for them to do." I've also read that Axiom subs don't put out like SVS or Hsu. I say that is a perfect indication those individuals have not optimized their systems for the exact reasons I stated above. Come to my place and listen to my subs growl with every note and flap your pants with every explosion. Having said that, I have no idea if I've optimized the growling and pant-flapping. And then it doesn't help the cause when I add in active LFRs and it all goes to hell precisely because I don't have the knowledge or technology to optimize my environment.

It's still a mystery to me why a 15 degree toe-in affected infrasonics from none to full. I'm actually not even sure about full. Maybe I can get more infrasonics out of my system with a 2 degree phase adjustment but how can I possibly know? I can spend the rest of my life tweaking and I frankly don't want to. I just want to enjoy sound...like I do with my measly M2OW system.

I will not make a post mentioning medication
I will not make a post mentioning medication
I will not make a post mentioning medication
I will not make a post mentioning medication

Aw crap, it slipped out. Sorry.

Seriously, it's a fair question. Every so often I spend a few days doing nothing but non-stop system tweaking and almost always end up with a significant improvement, but I probably won't be able to spend enough time to be confident I have an optimal setup until my hearing has degraded to the point where all I really need is a couple of midrange drivers anyways.

It is never-ending, and in some ways speakers like the LFRs are just another reminder that even the finest audio systems are arguably over-simplified relative to what we need to get everything perfect.

Even stereo imaging is a gross approximation of real world sound production, although I continue to be amazed by the uncanny realism it can provide, presumably because it all has to go through a pair of ears anyways.

Phase controls on subwoofers drive me crazy because they are a single control trying to deal with two separate variables:

- matching phase between subwoofer amp/wiring and main woofer amp/wiring (this is really a 0/180 degree thing)

- introducing delay to approximately compensate for the path distance between subwoofer/listener and main woofers/listener (two distances, one knob, get it right)

- I guess the phase control may also be compensating for different responsiveness between the sub and woofer drivers (heavier drivers accelerate more slowly) but I don't think so

It would be interesting to split the signal and run it through two separate phase controls, each one adjusted based on path distance relative to a different main woofer, then recombine the results. First thought is that it would make everything sound like "Rock On" , but I suppose that would not really be an improvement... the best you can do is get the subwoofer somewhere between the main woofers in terms of arrival time.

Yep. Phase knobs are really not the answer.

A phase knob can only retard the signal after arrival. Since the subwoofer requires advancement relative to the mains, it's a stop gap at best to simply phase align.

The solution is a DSP that can perform channel alignments relative to each other, not just after an input signal has arrived as in a subwoofer phase knob. This allows both phase AND cycle alignment.

Andrew, I suspected this in the past but now I'm sure. The rears need more 'nads. Either that or more angle. Or outboard drivers. Or all of the above. I'm sure the sound field can be more aethereal with more lobes.