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Bi-amping 101

Thank you Don,

Is DENON addressing your point (#6) in their user manual?
View attachment 149896

Yes. Whether or not bi-amping helps, or to what degree, depends greatly upon the output impedance of the amplifier, cable resistance (impedance), and speaker load (impedance). If the amplifier has low enough output impedance (high enough damping factor) and cables are large and/or short then it will prevent such modulation. The assumption for bi-amping in this case is that a single amplifier cannot handle it, the wires add too much resistance and allow signal modulation via back-EMF, and/or the crossover itself is allowing mixing. There are theoretical benefits but I remain highly skeptical of any real-world, audible, benefits. But I ain't tried it so what do I know...

The same argument applies to bi-wiring but with even less practical benefit since the only difference is whatever the wires themselves add. I have seen a few adverts that plot the current in each wire, showing how much less bass current is in the treble wire and vice-versa, but without recognizing either how much the interaction matters to the speaker nor how well the amplifier handles it (in the bi-wiring case the amplifier must still deal with all the current so any advantage depends on the wires, and higher-resistance wires leads to other problems, so...)
 
The phase mismatch from adding the Butterworth filters resulted in a deep suck out.

View attachment 149891

This is great, thanks for chiming in! Looking at the graph, I don't totally understand why that's happening? If the FMODs were both in-line to the same amp, then yes absolutely I'd expect a notch but because they're each attached to different amps I'm having a hard time imagining a suck out once the speaker starts pushing air?

I do not think so either, but he can try. The crossover will be completely mucked up unless the AVR compensates.

I probably did a bad job of framing this Q. I have no intention of doing this, I was simply trying to apply your teachings to my own equipment, simply because it's what I'm most familiar with, to see if I grasped the material. I absolutely agree, even if I didn't muck up the factory design, the benefits would be inaudible compared to the masking I have going on in the rest of my chain - a 16yo HTPC with 6 fans, untreated limestone/hardwood/glass listening room (not even carpet!), whatever Audyssey is doing in there because dad life doesn't give me time to investigate, etc etc. Likely just inaudible, period, given my average listening levels.

Even if lack of power were a concern, seems I'd be better off taking the 20 hours I'd spend calculating crossovers, running double cables, etc., and mowing lawns/shoveling driveways in the neighborhood for the $ to simply trade up for a bigger amp. Probably the best thing about being on this site for a few years is finding the huge tome of literature on the psychological impact of sighted listening, complexity bias, etc.

To answer your Q re: the AVR, I can't think of any way it could compensate because it works exclusively on a per-channel basis. I.E. it's just front left, no way to choose front left tweeter and/or front left woofer. Even Dirac does the same on every AVR implementation I'm aware of so I'd be looking at MiniDSP, or similar, in-line. Thanks again for the in-depth reply!
 
This is great, thanks for chiming in! Looking at the graph, I don't totally understand why that's happening? If the FMODs were both in-line to the same amp, then yes absolutely I'd expect a notch but because they're each attached to different amps I'm having a hard time imagining a suck out once the speaker starts pushing air?

I probably did a bad job of framing this Q. I have no intention of doing this, I was simply trying to apply your teachings to my own equipment, simply because it's what I'm most familiar with, to see if I grasped the material. I absolutely agree, even if I didn't muck up the factory design, the benefits would be inaudible compared to the masking I have going on in the rest of my chain - a 16yo HTPC with 6 fans, untreated limestone/hardwood/glass listening room (not even carpet!), whatever Audyssey is doing in there because dad life doesn't give me time to investigate, etc etc. Likely just inaudible, period, given my average listening levels.

Even if lack of power were a concern, seems I'd be better off taking the 20 hours I'd spend calculating crossovers, running double cables, etc., and mowing lawns/shoveling driveways in the neighborhood for the $ to simply trade up for a bigger amp. Probably the best thing about being on this site for a few years is finding the huge tome of literature on the psychological impact of sighted listening, complexity bias, etc.

To answer your Q re: the AVR, I can't think of any way it could compensate because it works exclusively on a per-channel basis. I.E. it's just front left, no way to choose front left tweeter and/or front left woofer. Even Dirac does the same on every AVR implementation I'm aware of so I'd be looking at MiniDSP, or similar, in-line. Thanks again for the in-depth reply!

The speaker is designed so that if you drive it with a single amplifier, or pair of amplifiers delivering the same signal (passive bi-amping using an AVR), then the drivers (woofer/tweeter) will "work together" to provide a cohesive wave front. That means getting the amplitude and phase correct from the speaker terminals, through the crossover, and out the drivers to your ears. When you stick additional filters in front of that, they are effectively in series with the crossover filters, and so change the filter characteristics. In particular, phase is changed by the extra filters as they roll off the signal, and that phase is not accounted for by the speaker itself. The result can be the woofer and tweeter are now out of phase, as @NTK showed very well, and you end up with a big suckout (dip, hole) at the crossover frequency. Nothing to do with the amps per se; it is because you have changed the signal driving the speakers to something they do not expect.

Your AVR should be able to analyze each channel and adjust the phase to fix that big dip. The catch is that, if it only looks at amplitude, it might not be able to properly correct the problem. The problem is the out-of-phase signals subtract instead of adding, and increasing the amplitude will not fix the null, just send more power into it. The analogy would be that 1 - 1 = 0, and if you amplify that, then 10 - 10 is still 0.

HTH - Don
 
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The top graphs are the responses using the standard LR4 with cross-over at 400 Hz. The bottom graphs are the responses when both the low-pass and high-pass filters are cascaded with 2nd order Butterworth filters with corner frequencies at 500 Hz.

The phase mismatch from adding the Butterworth filters resulted in a deep suck out.

Q about this - because this is a phase mismatch within a single enclosure, would the same suck out occur if I use the same crossover point between a booksehlf and sub? THat is, bookshelf = 80Hz HPF, sub = 80Hz LPF ?
 
Q about this - because this is a phase mismatch within a single enclosure, would the same suck out occur if I use the same crossover point between a booksehlf and sub? THat is, bookshelf = 80Hz HPF, sub = 80Hz LPF ?
The phase relationship in a bass managed system is already somewhat indeterminant because both the subwoofer and the bookshelves (plus their placements) have their own unknown phase shifts/delays. So phase mismatch is quite possible.

However, these can be readily rectified by measurements and room EQ, which is mandatory for optimal performance.
 
I actively bi-amp my "peak" speaker set. Current setup, which is my 2nd gen:
  • Digital source - hi-res music streaming or TV, mostly the former
  • DEQX HDP-5 (3-way digital bi-amp, selectable linear phase 192 db/octave crossover at 1270 or 1800 Hz on the 2-way speaker, crossed to sub L-R 48 db/octave at 80 Hz), I have an Earthworks M30 calibration mic, but I sometimes also like manually setting everything too.
  • 2 amp stack setups:
    • Accurate: 2 Purifi 1ET400A stereo amps, 1 for the mid/woofers, 1 for the tweeters
    • Sweet: Pass Labs X250.8 for mid/woofers, Pass Labs XA30.8 for the tweeters
  • Pair of sealed speakers I built with Purifi 6.5" X-version mid/woofers, Raal 70-20XR ribbons, drivers directly connected to the binding posts.
  • Rythmik 15" sealed sub
It sounds Freakishly accurate in "accurate mode", and a bit less so in "sweet mode" but super-engaging, and lots of fun to listen to different music on with either. I also have a LTA Microzotl tube proamp I can insert in the chain before the DEQX which messes with the sound character too, which is also fun sometimes.

My first active bi-amping experiments started with a set of custom speakers I'd bought from someone else with excellent drivers but terrible crossovers. I tried to alter the analog crossovers a few times and was never happy with it. So for fun I removed the crossovers and tried active bi-amping using a Behringer DCX2496, it was a bit (but notably) better, and I have successively improved different parts of the chain until it's now much better.

I'm now working on my 3rd gen setup with a bit better and lower distortion drivers, changing out the tweeter for a Raal 140-15D (goes a bit lower and has a bit lower distortion in the lower range... it goes higher too but I'm sure nobody can hear the difference there!), and the mid/woofer for a pre-production Purifi X-series aluminum mid/woofer (lower harmonic and IMD across maybe even a bit more range). I suspect I will not be able to subjectively tell the difference if I set the configuration the same way, but it will be possible to cross over higher or lower than before experimentally, so that's kind of fun.

My 2nd gen setup is likely already at (or beyond) the limit of my ability to detect a difference subjectively. We'll see in maybe 2 months or so when the 3rd gen is all together.
 
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Hey.

I'm bi amped . I have a marantz 7005 and a bk205.5 amp. I run a asend acoustics center and vandy 3.5 sigs . I think when I went from non biamped to biamped I heard a big difference in the bass mostly but now it's been so long I think I would need to switch back to think if I really do hear a difference.

The more I read about audio and listen the more I think I'm confused about what I think I hear and what I'm hearing.

This God dam sport is a pain in the ass.

But I love it.

...I have been reading this and it is so far over my head I am just giving up and putting on some tunes...

...I did note the same "differences" between two dacs...when I tried a new DAC I was pretty sure there was a rolled off treble, added reverb, and boosted base...

...thing is, me gold ear noted the same effect when I switched back...

...amirm noted this phenomena in one of his many articles..when you put new stuff in play you really listen...
 
Sorry, a totally newbie question here: how does the speaker "know" which frequencies to suppress and which one to process? In a passive bi-amping situation, if entire spectrum of frequencies are passed to the speakers, how does one speaker "know" they have to suppress say, Lower Frequencies and the other speaker will take care of the LF?
 
Sorry, a totally newbie question here: how does the speaker "know" which frequencies to suppress and which one to process? In a passive bi-amping situation, if entire spectrum of frequencies are passed to the speakers, how does one speaker "know" they have to suppress say, Lower Frequencies and the other speaker will take care of the LF?

This other thread on ASR has more content about active vs. passive bi-amping: Bi-amp, bi-wire, passive crossover & active crossover integration. However, I'll give a really short intro since even there it doesn't seem to describe the basic idea which looks like where your confusion is.

You can only usefully do "passive bi-amping" with an existing analog crossover, and the analog crossover does that for you. The key is that it's really NOT rejecting anything from the amplifier, it's simply not drawing power for the frequencies that are not useful for the speaker element.

So, looking at just one of the speaker elements, say a tweeter, the crossover for that tweeter is increasing the impedance to the amplifier for sound frequencies below the "crossover point". The voltages sent by the amplifier for the full-range sound are the same, but the increased impedance at the lower frequencies mean that it is drawing basically negligible power below that "crossover point".

Say you have a 2-way speaker with a tweeter for high frequencies and a mid-woofer for low frequencies, like a bookshelf kind of speaker. Often they have 2 sets of binding posts on the back with metal straps connecting the left-hand ones together and the right-hand ones together. There is a high-pass filter on the inputs for the tweeter, and a low-pass filter on the inputs for the mid-woofer. Together they cover the whole frequency range (more or less).

Now, consider disconnecting the 2 sets, then using 2 amps, one connected to each set, both using the same full frequency range. Well, because of the analog crossover/filters, each amp will effectively only be providing power for one of the frequency ranges. Yes, again, the full voltage swing on all frequencies are expressed, but the current and power will only be drawn for the unfiltered part of the frequency range on each amp.
 
I wonder If Amir ever did a speaker test with the klippler when the speaker was biamped and if there was a difference when it was not? I don't think I remember reading that before.
 
I wonder If Amir ever did a speaker test with the klippler when the speaker was biamped and if there was a difference when it was not? I don't think I remember reading that before.

Running the numbers shows the effect is pretty small unless you have an active crossover, which usually can do integrated DSP correction, at least for the active crossovers that are in the digital domain.
 
So, basically not much value in spending extra to get another (same) power amp in order to segregate Highs/mids and lows that could've *potentially* contributed to the sound improvements (as per my dealer :) )
 
Sorry, a totally newbie question here: how does the speaker "know" which frequencies to suppress and which one to process? In a passive bi-amping situation, if entire spectrum of frequencies are passed to the speakers, how does one speaker "know" they have to suppress say, Lower Frequencies and the other speaker will take care of the LF?
As @eboleyn said it is up to the crossover inside the speaker. The crossover is a combination of a high-pass filter that lets highs through to the tweeter and low-pass filter that lets lows go to the woofer, block the highs from the woofer (and lows from the tweeter). Look at the diagram in the first post.

HTH - Don
 
I wonder If Amir ever did a speaker test with the klippler when the speaker was biamped and if there was a difference when it was not? I don't think I remember reading that before.

The Klippel NFS is measuring the speaker's acoustic output, and only the speaker's acoustic output. It doesn't have anything to do with bi-amping or not. It doesn't measure the amp, or the amp's influence on the speaker, or the speaker's influence on an amp.
 
This article provides a quick look at how bi-amplification works. The conventional scheme is to split the signal into two frequency bands before the power amplifiers that drive the speakers. The bass (LF) amp sees only LF signals and drives only the LF driver (woofer). The treble (HF) amp sees only HF signals and drives only the HF driver. This reduces the signal for each amplifier, improving headroom and potentially allowing amplifiers to be chosen for their target frequency range without the burden of supplying the full-range audio signal.

Here is a regular single-amplifier system and passive speaker crossover. The crossover comprises a low-pass filter (LPF) for the woofer and high-pass filter (HPF) for the tweeter. The crossover is inside the speaker; the dashed outline shows there can be a single or split (high/low) set of speaker input terminals, but the connection is the same as far as the amplifier is concerned.

View attachment 126273

For this experiment both crossovers are second-order (12 dB/octave) Linkwitz-Riley designs and the same whether passive or active. The crossover frequency is 1 kHz. The speakers are represented by 8-ohm resistors in this simplified analysis. This is the transfer function (frequency response) of the crossover:

View attachment 126274

For analysis I am using two signals at 300 Hz (LF) and 3 kHz (HF) which on a log scale sit on either side of the crossover:

View attachment 126275

The output voltage from the amplifier includes both signals, of course:

View attachment 126276

You can see how the low-frequency signal and high-frequency combine to create the modulated output signal. Since both signals have the same amplitude, the combined signal is twice the level of the individual signals. While this makes it easy to see, note in practice HF signals are typically much lower in amplitude than the LF signals (see Equal Loudness Curves in Wikipedia or wherever).

The relative voltage, current, and power is in the table below. The numbers are low since I am using a small signal; everything is relative. The combined RMS voltage for this case is simply sqrt(LF^2 + HF^2). The power is the average power, the product of RMS voltage and current (RMS power is a meaningless term though often used mistakenly).

View attachment 126281

Now repeat the experiment, but place a line-level crossover before the power amps. The crossover is typically active but could be passive; the usurpation of the term “passive bi-amplification” the way AVR marketing defines it is an on-going source of confusion.

View attachment 126282

There are now two amplifiers but each only handles a portion of the signal divided into two (low and high) frequency bands. Since the crossover is not ideal, that is does not drop instantly to zero on either side of the crossover frequency, a little bit of the HF signal is still seen in the LF output and vice-versa. It is a little hard to see in the time-domain plot, but the frequency-domain plots clearly show this.

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Now we can create a new table showing the parameters of each amplifier in the bi-amplified case:

View attachment 126286

The voltages are similar to the previous table except now they are at the output of the amplifiers and, with the filters, the RMS value is a little lower. The RMS current is reduced more significantly since the load is split between the two amplifiers, and power is also significantly reduced (by about 60%). Note power is not a linear function, but rather the product of voltage and current (a nonlinear function), so splitting the signal this way does not reduce the power by exactly one-half (it is better than that). You could use two amplifiers each one-half the rated power if the single amplifier and deliver the same power to the speaker. The amplifiers also directly drive the speakers so there is no crossover loss, and no chance of the signal from one speaker modulating the other through the crossover circuit, potentially reducing distortion.

Now consider “passive” bi-amping as implanted by most AVRs and shown below. In this scenario both amplifiers are driven by the same signal, so their outputs have exactly the same voltage. There is no benefit in voltage and, since most amplifiers are essentially voltage-mode designs, no increase in system headroom. Current is reduced from each amplifier, since the passive crossover in the speaker “blocks” current that is out of band, and there is a corresponding reduction in power (not as much as when an active crossover is used, however).

View attachment 126287

The voltage output of each amplifier is the same:

View attachment 126288

The current is lower since the load seen by each amplifier is only part of the speaker. The LF amp has almost no HF current in its output, and similarly the HF amp has little LF current, even though the voltage is the same.

View attachment 126289

Here is the parameter table for the “passive” bi-amping case:

View attachment 126290

And a comparison of all three examples:

View attachment 126291

A ratio less than one represents an advantage (decrease) in the parameter the amplifier is delivering to the speaker. In a conventional active bi-amp system, each amplifier delivers only 65% of the voltage and current and only 42% the power of a single amplifier providing the same signal to the speakers. A “passive” bi-amp system the way an AVR does it has no voltage benefit but does require less current from each amplifier, resulting in the same 65% reduction in current, but due to the higher voltage must deliver 65% of the power. A reduction, true, but less dramatic than for an active bi-amp system, and since there is no reduction in the voltage that must be delivered there is no (voltage) headroom advantage. Where an active system would allow you to replace a 200 W amplifier with two 100 W amplifiers, the passive scheme requires two 200 W amplifiers to maintain the voltage headroom required.

One other significant consideration is how we hear as reflected in the equal-loudness curves (see e.g. https://en.wikipedia.org/wiki/Equal-loudness_contour). It take much higher levels of low-frequency signals to sound as loud as midrange signals. At 80 dB SPL, a 100 Hz signal must be about 10 dB louder (10x the power) to sound as loud as a 1 kHz signal, and must be about 20 dB louder (100x the power) by around 60 Hz. That is one reason bi-amplified systems (of the non-passive kind) often use a much larger bass amplifier than for the tweeter.

A few key points about “passive” bi-amping, from the original version of this roughly ten years ago:
  1. Each amplifier requires the same voltage output as a single amplifier since they each have the same signal. There is no voltage headroom benefit.
  2. Because the speaker load is essentially an “open” in the unused frequency band, less current output is required from each amp.
  3. There is no net system power increase at the speakers assuming the amps have the same voltage rails (e.g. inside an AVR or multichannel amplifier with the same power voltage rails to all amps). If you had a 100 W amp before, and bi-amp with two 100 W amplifiers, passive bi-amping does not give you 200 W to the speaker. You have split the load into two frequency bands, but the maximum power is the same to the speaker. That is, 100 W to the lows and 100 W to the highs is the same as having a 100 W amp that covers the entire frequency range. It is not the same as driving the speaker with a 200 W amplifier; to increase the power, you need to increase the voltage rails. There is not an effective increase in power headroom as there is for an active approach.
  4. In fact, there is more power lost, since the amps are not 100% efficient. That is, it actually take more energy from the power supply to passively bi-amp than if you used a single amp. This is also true for active bi-amping, but in that case we can choose lower-power amps for the highs (which rarely need the same power as the lows) and realize net power savings. That does not happen with (typical) passive bi-amping.
  5. There is no damping factor improvement over a single amp since the speaker crossovers are still in-circuit. One of the benefits of active bi-amping is direct connection from amp to driver, providing better driver control; this is not true in passive bi-amping.
  6. There is no longer electrical interaction among drivers with passive (or active) bi-amping. (There may still be mechanical coupling if the drivers are not isolated from each other.) That is, if the woofer starts to distort the input signal through electromechanical forces, it no longer modulates the HF amp’s output. One plus for bi-amping, active or passive.
  7. If the amps share a power supply, as do most AVRs and many (most?) multichannel amps, then modulation between high and low amps can still occur through the power supply. This can also happen with active bi-amping, although separate amps are the norm in the pro world. At least when I have done it…
  8. There may be some distortion reduction since power output is lessened in the amps. I suspect this is not significant, but it should happen due to the lower current draw. The catch is that the voltage swing of each amp is unchanged, so any distortion related to voltage swing is not changed. Only distortion components depending on output current may be reduced. That is design-dependent, but since most amps are primarily voltage-mode amps, I suspect any distortion reduction is small.
  9. You have two amps now so presumably noise is a little higher since you have two uncorrelated noise sources. At the speaker outputs I suspect it’s a wash since only a reduced frequency band gets through the drivers to hear.
  10. Thermally it is a loss since no amp is 100% efficient. There is always a little “waste” power that gets turned into heat, both standing bias current (especially if not class D amps) and losses through the components in the amp. Thus passive bi-amping will cause your AVR/amp to run hotter than if using a single amp (assuming unused channels). It is worth noting that amplifiers are typically most efficient at maximum output; the HF amp is probably loafing most of the time and thus wasting power and generating heat.
So, there are some potential benefits using passive bi-amping, but I suspect they are inaudible (I have not tried passive bi-amping so cannot say). And a lot of drawbacks. The major benefit is mostly mental, IMO; users can now use their “extra” amp channels. Whether this benefits anyone other than the electric company I cannot say, but I suspect not… It does eliminate signal “bleed” through the crossover, so if the amplifier has high output impedance (like a tube amp) then passive bi-amping could benefit. If the amplifier is near clipping, or current-limited, then the reduction in current by a passive bi-amp scheme could reduce distortion. Again I suspect that is an insignificant improvement.

FWIWFM - Don
I occasionally think of how to test one aspect of "passive" bi-amping, the potential improvement in distortion you mention in point #6. At the risk of opening up a can of worms...

If a multitone or sweeps are fed into a bi-ampable speaker, perhaps the alleged distortion from the woofer can be measured in the tweeter's output. In this case a Paradigm Export Monitor 2-way with an 8" woofer. Big woofer for a 2-way, I am thinking this may be good or bad case study...
I am using monoblocks, so the shared power supply / chassis variable is avoided.
I close-mic the tweeter while driving the speaker with various sweeps and multitone signals with the goal of causing woofer distortion that can have an impact on the measured output of the tweeter when a single amp is used...
1699588846253.png


I set up a the speaker so I can swap configurations without moving the mic or changing the gain or any external factor in any way.

1699588910489.png


First, do some sweeps:
I do one with bi-amp (aka 'Forward'), then I swap the amps between tweeter and woofer (aka 'Reverse') The allows control for any channel to channel gain differences. Then I sweep with the speaker single-amped (orange trace).
1699589181135.png

Zooming in:
1699589435791.png

The gain is matched acceptably, no frequency response differences between configurations. This is a good result, and simplifies the distortion comparisons.

Comparing distortion as measured at the tweeter:
1699591527687.png



I see no difference between bi-amp and single amp while doing these fairly high power sweeps. I tried stepped sine tones to try to increase the sensitivity. It didn't change the outcome. I was worried about warming up the drivers at the volumes, but didn't notice that in the data.

I tried different multi-tone signals driving the woofer as hard as I could, for instance 60Hz and 3500Hz.
1699590255146.png


I see no difference in distortion between amplification configurations.

Maybe I need to turn it up to eleven, risk melting a voice coil, in order to see a measurable improvement with passive bi-amping. But in that case, I likely need a more efficient speaker.;)
 
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I occasionally think of how to test one aspect of "passive" bi-amping, the potential improvement in distortion you mention in point #6. At the risk of opening up a can of worms...

If a multitone or sweeps are fed into a bi-ampable speaker, perhaps the alleged distortion from the woofer can be measured in the tweeter's output. In this case a Paradigm Export Monitor 2-way with an 8" woofer. Big woofer for a 2-way, I am thinking this may be good or bad case study...
I am using monoblocks, so the shared power supply / chassis variable is avoided.
I close-mic the tweeter while driving the speaker with various sweeps and multitone signals with the goal of causing woofer distortion that can have an impact on the measured output of the tweeter when a single amp is used...
View attachment 325092

I set up a the speaker so I can swap configurations without moving the mic or changing the gain or any external factor in any way.

View attachment 325093

First, do some sweeps:
I do one with bi-amp (aka 'Forward'), then I swap the amps between tweeter and woofer (aka 'Reverse') The allows control for any channel to channel gain differences. Then I sweep with the speaker single-amped (orange trace).
View attachment 325094
Zooming in:
View attachment 325096
The gain is matched acceptably, no frequency response differences between configurations. This is a good result, and simplifies the distortion comparisons.

Comparing distortion as measured at the tweeter:
View attachment 325105


I see no difference between bi-amp and single amp while doing these fairly high power sweeps. I tried stepped sine tones to try to increase the sensitivity. It didn't change the outcome. I was worried about warming up the drivers at the volumes, but didn't notice that in the data.

I tried different multi-tone signals driving the woofer as hard as I could, for instance 60Hz and 3500Hz.
View attachment 325101

I see no difference in distortion between amplification configurations.

Maybe I need to turn it up to eleven, risk melting a voice coil, in order to see a measurable improvement with passive bi-amping. But in that case, I likely need a more efficient speaker.;)
it almost looks like george lucas THX sound system crossover how did you achieve this , yet can it do CIC Empire Leicester Square at star trek 6 opening music at THX levels

Screenshot 2023-11-18 07.27.29.png
376922839_10160833475435149_7034826268496599410_n.jpg
793684_528842240489508_1607986349_o.jpg
 
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it almost looks like george lucas THX sound system crossover how did you achieve this , yet can it do CIC Empire Leicester Square at star trek 6 opening music at THX levels

View attachment 327104
Hi Andysu, George Lucas motivates me to do lots of things.
But no, this speaker cannot, and would disappoint trying to come even close to your expectation!
Peace!!!
 
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