Bi-Amping Question

[Slave IV]

Not mine, I just linked to it since I have used it a lot. The equations are well-known but I am too lazy to dig up my old graduate accoustics book and look them up again.

"Damping"; "dampening" is about getting wet.

We are fairly insensitive to distortion in music, depending on the type of distortion, and unless something is broken the speakers are usually orders of magnitude higher in distortion than the electronics.


Speakers and room most certainly have the biggest influence on the sound, with amplification a distant second unless the amplifier is not up to the task. An underpowered amplifier that is clipping, one with high noise level, or one with high enough output impedance that it changes the expected frequency response of the speakers are common issues when folk really do need a different amplifier.


The answers are usually (should be) qualified by all sorts of stuff I and others have been saying. It is not just power, but the amount and type of distortion, noise level, output impedance, input sensitivity/gain, and less tangible things like build quality, warranty, and so forth all factor into choosing an amplifier. IMO there is a bias here for performance, such as Benchmark's AHB2 and similar amplifiers that exhibit outstandingly low distortion and noise, that clearly outperform any AVR I have read about. But how much of the difference is audible is up for debate; many people would be astounded to discover how little difference in sound there is among competent amplifiers driving typical speakers at levels well within the amplifiers' power range. But there are always exceptions, difficult speaker loads, noisy amplifiers, etc. that lead to marketing that "everybody" has systems that are above average and represent extreme cases requiring only the best (or at least most expensive) amplifiers on the planet.

In practice, the answer to your last question is "yes" more often than people think, even among amplifiers considered widely disparate like tube and solid-state designs (not all tube or SS amps, natch). But I have added amplifiers to an AVR that was clearly not up to the task of driving an array of inefficient low-impedance speakers, and also experienced the case of finding no difference between the sound of a different AVR driving the same speakers with and without an amplifier. Easily measurable in both cases. If one amplifier produces (or is rated for) the same power but >1% distortion whilst the other is delivering the same power at 0.1%, you may tell the difference. Similarly if one amplifier's output impedance is a few ohms (e.g. a tube amplifier) and the other is a SS amp with output impedance <0.1 ohms, again you may be able to tell a difference in frequency response for certain speakers. The problem is figuring out what we actually hear versus what we think or what to hear, and that is where controlled testing becomes so important. I have on many occasions been sure of the difference between two components, only to discover there was none I or others could hear in a blind test. And almost as often discovered differences I did not expect among similar components. Life goes on.

Whatever - Don

lol! I thought something was off with my autocorrect phone typing but obviously wasn’t thinking hard enough on that part.

This all makes sense and it’s all definitely making me more at ease when it comes to amplifier selection in the future. I know it’s hard to say how much of a difference people can hear and how much it is worth to pay more for an amplifier. I still believe it does make a difference but keep in mind, I didn’t go out and pay thousands of dollars for an amp, which before too long ago, I thought was the only option if I wanted a high quality external amplifier to run my system. Before I did this test with my receiver, I did do some proper a/b testing in another room with another amp and a switcher where I matched levels and was able to seamlessly switch between two different amp and two different speaker setups. To me, the sound differences were so subtle, I could barely tell the difference but there was still something changing. When I brought in my blind tester, they knew nothing other than sitting there listening and they were able to describe changes as soon as I switched anything. Their observations were in line with what I thought could be the differences and their preferences were different depending on the type of music I was playing and these preferences all made sense to me based on which setups should be more ideal for the respective types of music I was playing. I would say that overall, the preference was mostly for the other amp, that is rated at under half the power output of the monoblocks but running with a tube preamp. So anyways, I am convinced that the monoblocks are an upgrade over my receivers internal amps, enough so that they are worth the few hundred dollars they are. I’m also convinced I probably wont ever need to spend thousands on a mega-powered, prestigious name brand amplifier. I’m pretty happy where I’m at now with what I have and will probably pick up one more matching or similar amp down the line so my front three can all be externally powered but no hurry for that. As I said earlier, I’m not going to mess around with this stuff too much anymore until I can commit to a fully active speaker system.

 

[jhaider]

I'm experimenting with bi-amping and amplifiers in general.

Quick background is I got a couple of monobock amplifiers to start replacing channels being run from my receiver. The monoblocks have slightly higher power rating at 8 ohms, which is what my front and rear surrounds speakers are rated at. My first test using the monoblocks to run the front mains resulted in very positive and noticeable difference. My receiver has Audessy and it was already calibrated using all internal amps and I did not change any settings when switching to external monoblocks, just used the RCA pre-outs for the channels and swapped the speaker wire over. The output wasn't noticeably louder but the soundstage, clarity and everything else I could think of was improved.

Next test and this is where my question comes in is when I tested the monoblocks on the center channel. The center channel is matched from the same brand and line as the mains but it is rated at 4 ohms and my receiver is not rated to run 4 ohms but it has been fine in that configuration. The monoblocks are rated to run 4 ohms and should be delivering more than double the power the receiver could at that load. First thing I did was just use one monoblock hooked up to the center speaker same way I did the mains, by using the RCA preout and swapping the wires over and no changes to any settings. My expectation was that the difference would be even more noticeable but what resulted was lower volume and clarity. This is already very odd to me and makes no sense considering the results I got earlier with the fronts. So, I then used an RCA Y splitter cable to feed the center preout to both monoblocks and added wiring to bi-amp the center speaker. Now, it would be getting more than 4 times the power between the tweeter and mids and surely be a huge difference. Only thing is it ended up sounding about the same as just the single monoblock running the center and still less loud and clear as just using the receiver's internal amp. I haven't had time to test any further and just went back to running the front mains off the monoblocks since that had the best results but what could be causing this issue?

My first thought is that using the Y cable should be reducing the input signal to each amp so I need to adjust the gains but I was surprised that I did not see any mention of this when I was searching for a solution. Has anyone else had this issue and does this sound like it could be the culprit? I haven't done any math other than the basic understanding that the power ratings alone should have resulted in a higher output in both center channel tests regardless of gain settings since everything else stayed the same between all these tests. Am I missing anything? Thanks for any insight on this as it is my first post here and my first time dealing with monoblocks and bi-amping.

Recalibrate (and repair the damage Audyssey does to the in-room response curve) and you’ll be fine. All you’re hearing is differences in gain structure.

Or take the new amps out then you’ll be fine too.

 

[Slave IV]

Recalibrate (and repair the damage Audyssey does to the in-room response curve) and you’ll be fine. All you’re hearing is differences in gain structure.

Or take the new amps out then you’ll be fine too.

It was a gain issue and it should have been obvious to me but I was trying to figure out if it was caused by something else without considering the preouts would have different levels (still don’t get why they would design it like that).

It is really interesting to me that the reoccurring advice here seems to be that amplifiers are basically worthless though. It sounds like no one here thinks there is any difference in sound produced by any decent receiver, separate component setup, integrated amp or basically any kind of amp. So is everyone here running thousand dollar plus speakers and running them off their TVs speaker outs because your speakers are so efficient they don’t need anything more? I’m totally joking and respectful of all the input here but what’s being suggested here doesn’t sound too much off that example. I am seriously curious what kind of amps people here are running now though.

 

[jhaider]

It was a gain issue and it should have been obvious to me but I was trying to figure out if it was caused by something else without considering the preouts would have different levels (still don’t get why they would design it like that).

Nothing to do with preout levels, but gain structure differences between internal amps and external amps.

It is really interesting to me that the reoccurring advice here seems to be that amplifiers are basically worthless though.

That’s a misread. Amps matter when you need the extra headroom or have difficult to drive speakers. Or when they have extra functionality, such as integrated signal processing, and you know how to use it.

I’m totally joking and respectful of all the input here but what’s being suggested here doesn’t sound too much off that example. I am seriously curious what kind of amps people here are running now though.

Well, speaking only for me in one system I’m driving Revel M126Be with base model Anthem AVR - MRX 520? I think one was measured here and may have earned headless panther. It’s fine. Another uses a small Lyngdorf integrated to drive Stereophile Class A (limited LF) rated speakers (Technics C700). The room correction is great, the amp is good enough.

The others use separate preamps, due to the processing I want and/or use of powered speakers or speakers with hybrid passive+DSP crossovers. So no choice to use AVR amps.

 

[Doodski]

It was a gain issue and it should have been obvious to me but I was trying to figure out if it was caused by something else without considering the preouts would have different levels (still don’t get why they would design it like that).

It is really interesting to me that the reoccurring advice here seems to be that amplifiers are basically worthless though. It sounds like no one here thinks there is any difference in sound produced by any decent receiver, separate component setup, integrated amp or basically any kind of amp. So is everyone here running thousand dollar plus speakers and running them off their TVs speaker outs because your speakers are so efficient they don’t need anything more? I’m totally joking and respectful of all the input here but what’s being suggested here doesn’t sound too much off that example. I am seriously curious what kind of amps people here are running now though.

It's not that all amps sound the same but quality well designed uncolored amps sound the same when operated within their capability. Your TV speaker outputs probably don't sound as good as a proper amp but maybe up to a couple watts a channel it might.

 

[Wolf11Man]

The (usually small) advantages of bi-amping are even lower with systems using a subwoofer, which is the norm for HT systems.
The lowest frequencys, that may stress the amp with a normal stereo system, are not present when the sub takes over at 80 Hz.

A fully active system is a completely other animal, as it basically needs a reconstruction an new development of the speaker.
Which means removing the passive crossover and reconstructing it on the active domain.

On the other hand and this has not been said so far, there is no disadvantage in bi-amping at all, if levels are set correct.

Maybe some other idea with all these amps?
With my 5.2.4 HT set up I use separate amps for any speaker except R, L and Center. This, in theory, will give me more headroom for the AVR's internal amps which have a power supply that is much too small to drive the 7 internal amps to full power. Just compare an AVR's 2-channel power output and what is left of it, all channels driven. The impressive 2x180 Watt often crumble to wimpy 7x 45 Watt or the like. Most manufacturers don't give this number for a reason.

Is it audible? I really don't know, because I can not do a A-B comparison, but it sure doesn't sound worse. You have to run Audyssey again, with the new amp configuration, of course.

 

[Slave IV]

Nothing to do with preout levels, but gain structure differences between internal amps and external amps.



That’s a misread. Amps matter when you need the extra headroom or have difficult to drive speakers. Or when they have extra functionality, such as integrated signal processing, and you know how to use it.



Well, speaking only for me in one system I’m driving Revel M126Be with base model Anthem AVR - MRX 520? I think one was measured here and may have earned headless panther. It’s fine. Another uses a small Lyngdorf integrated to drive Stereophile Class A (limited LF) rated speakers (Technics C700). The room correction is great, the amp is good enough.

The others use separate preamps, due to the processing I want and/or use of powered speakers or speakers with hybrid passive+DSP crossovers. So no choice to use AVR amps.

Not sure is you read everything since there is already a lot here but it seems like it did have something to do with preout levels unless you are implying the internal amplifiers are all setup differently. The same test when using the external amps to replace the front left and right channels did not have such a decrease in output level compared to the center, which is also a more sensitive speaker. Either way with that, the resulting fix was just adjusting the gain up a bit when using the external amps on the center speaker.

And yes, I was taking into account that the amps have what it takes for the intended job but what I’m hearing is that is the only requirement is that so you should basically buy the cheapest amp possible that meet your power or feature requirements.

Thanks for sharing some of your setups with reasoning behind them. That all makes good sense to me.

It's not that all amps sound the same but quality well designed uncolored amps sound the same when operated within their capability. Your TV speaker outputs probably don't sound as good as a proper amp but maybe up to a couple watts a channel it might

Right, I’m always assuming these examples are for amps matched to speakers that suit their capabilities. Hence the joke example of built in TV amp possibly be good enough if you are running extremely efficient speakers.

The (usually small) advantages of bi-amping are even lower with systems using a subwoofer, which is the norm for HT systems.
The lowest frequencys, that may stress the amp with a normal stereo system, are not present when the sub takes over at 80 Hz.

A fully active system is a completely other animal, as it basically needs a reconstruction an new development of the speaker.
Which means removing the passive crossover and reconstructing it on the active domain.

On the other hand and this has not been said so far, there is no disadvantage in bi-amping at all, if levels are set correct.

Maybe some other idea with all these amps?
With my 5.2.4 HT set up I use separate amps for any speaker except R, L and Center. This, in theory, will give me more headroom for the AVR's internal amps which have a power supply that is much too small to drive the 7 internal amps to full power. Just compare an AVR's 2-channel power output and what is left of it, all channels driven. The impressive 2x180 Watt often crumble to wimpy 7x 45 Watt or the like. Most manufacturers don't give this number for a reason.

Is it audible? I really don't know, because I can not do a A-B comparison, but it sure doesn't sound worse. You have to run Audyssey again, with the new amp configuration, of course.

Definitely agree and one of the goals I had was lightening the load for my avr and allowing it to run less channels. Particularly the center since my AVR isn’t rated for 4ohm speakers even though I’ve never experience an issue with running an amp at higher than rated loads in any situation in the past. Another thought I had was using one of the external monoblocks to run an extra subwoofer I have lying around. It is for one of my cars but I haven’t got around to setting it up so it’s just sitting in the garage. Not sure if it would be worth the trouble since it would be mismatched with my existing powered subwoofer but maybe another fun experiment for later on.

 

[MaxwellsEq]

Are you saying there should be no difference between a receiver, running separates, or different amplifiers as long as they are producing similar power under similar loads?

You don't have to take anyone's word for what gets written here. The spirit of ASR includes testing for yourself. Firstly, you must, must, must match levels to 0.1V at the speaker terminals. Secondly you need make the test in such a way that you can't see (or know via any other "tell" such as click, or buzz) which amplifier you are listening to - which means someone else has to do the swapping for you. Thirdly, whoever is doing the swapping must ALSO not know which amplifier is which, because even if they don't want to, they will signal to you in some manner which is which. Fourthly, you have to run the test at least 10 times.

If that sounds like a lot of effort, it is, which is why most people don't bother. But if they do, they discover that (assuming the amplifiers are not outside their comfort zone, measure flat, are not too noisy or distorting), and assuming that the levels are identical, they can't tell them apart. It turns out, many of the things reviewers hear and we hear vanish when the levels are exactly matched.

 

[dualazmak]

If you want multiple amps and to divide up the power to the various drivers then go with an active crossover and use separate amps for the drivers. Bi-amping is a very very small difference in a passive crossover system that is not really audible but it has certainly taken over folklore and snake oil science.

I fully agree with you!

In multichannel setup of "active crossover and use separate amps"(eliminating all the passive LCR-network and attenuators; SP drivers are dedicatedly and directly driven by each amplifier), i.e. in PC-DSP-based multichannel multi-SP-driver multi-amplifier audio setup, you can select amplifiers in policy of "right-person-in-right-place".

For example, we do not need powerful amplifiers to drive our tweeters and/or super-tweeters, but we need high S/N low distortion small power amplifiers in these places; please refer to my posts;
- Even Greg Timbers uses "reasonable and budget" Pioneer Elite A-20 for compression drivers (super tweeters) in his extraordinary expensive multichannel stereo system with JBL Everest DD67000 which he himself designed and developed: #435
- (Provisional) Decision on amplifiers selection and photos of the listening environments: #311

If you OP @Slave IV would be further interested, please find details of my latest multichannel multi-SP-driver multi-amplifier audio system in my post #931 on my project thread;
- The latest system setup of my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig, including updated startup/ignition sequences and shutdown sequences: as of June 26, 2024: #931

 

[Slave IV]

You don't have to take anyone's word for what gets written here. The spirit of ASR includes testing for yourself. Firstly, you must, must, must match levels to 0.1V at the speaker terminals. Secondly you need make the test in such a way that you can't see (or know via any other "tell" such as click, or buzz) which amplifier you are listening to - which means someone else has to do the swapping for you. Thirdly, whoever is doing the swapping must ALSO not know which amplifier is which, because even if they don't want to, they will signal to you in some manner which is which. Fourthly, you have to run the test at least 10 times.

If that sounds like a lot of effort, it is, which is why most people don't bother. But if they do, they discover that (assuming the amplifiers are not outside their comfort zone, measure flat, are not too noisy or distorting), and assuming that the levels are identical, they can't tell them apart. It turns out, many of the things reviewers hear and we hear vanish when the levels are exactly matched.

This method definitely seems thourough. My only question about it is how much fatigue comes into play. After 10+ tests, depending on how long, what the content is and what levels, people could just be over it and not want to hear a difference anymore just to get it over with, lol!

I’ve been very unsure about the differences myself and question my own critical hearing abilities so I listened to several comparison tests online where I believe the levels were matched and I had no idea what order the devices were being played. It would just cycle through with various content and each device was played well over 10 times. I could say that every time, I was able to at least decide on a favorite and that favorite sound I heard corresponded with what the intended description of that sound should have been after the test. Some of these tests even included the same amplifier compared with itself with different op-amps. So even if I was somehow fooling myself with some sort of bias, I don’t know how that bias could have consistently selected correctly across multiple tests where the order was changed and I had no idea what that order was. Especially when all the samples should sound exactly the same according to what is being said here. I’m not trying to negate what you are saying and actually want to confirm it myself but I’m just reporting the results I’ve had with my recent focus on very extensive testing at least for me.

This topic still intrigues me and I will attempt more testing when I am able to recreate the parameters you laid out but for now, I can say that I am enjoying my sound systems more now that I’ve gone through the recent rigorous process of testing all the variations. The subtle characteristics that were bothering me before I added the external amps to my center channel are no longer present although I feel I’m now more able to pick out the subtle characteristics that bother me about the speaker itself. I’m a manner, I would say the external amps have made the sound more transparent to me and I do get there could be bias involved in this case but whatever it is, I’m enjoying my content more without thinking about what’s going into it, which I think is a big part of what we are trying to achieve with this audiophile madness.

I fully agree with you!

In multichannel setup of "active crossover and use separate amps"(eliminating all the passive LCR-network and attenuators; SP drivers are dedicatedly and directly driven by each amplifier), i.e. in PC-DSP-based multichannel multi-SP-driver multi-amplifier audio setup, you can select amplifiers in policy of "right-person-in-right-place".

For example, we do not need powerful amplifiers to drive our tweeters and/or super-tweeters, but we need high S/N low distortion small power amplifiers in these places; please refer to my posts;
- Even Greg Timbers uses "reasonable and budget" Pioneer Elite A-20 for compression drivers (super tweeters) in his extraordinary expensive multichannel stereo system with JBL Everest DD67000 which he himself designed and developed: #435
- (Provisional) Decision on amplifiers selection and photos of the listening environments: #311

If you OP @Slave IV would be further interested, please find details of my latest multichannel multi-SP-driver multi-amplifier audio system in my post #931 on my project thread;
- The latest system setup of my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig, including updated startup/ignition sequences and shutdown sequences: as of June 26, 2024: #931

Wow! This is some incredible detail and dedication! Thanks for sharing! I haven’t gone through all of it closely yet but I noticed you are crossing over your subwoofer feed with a 12db/octave at 50hz and additionally with the subwoofer’s built in crossover at 55hz 24db/octave. I’ve done similar myself in the past with some setups but I’m curious as to what the specific intention of doing this in your case is, if any.

 

[MaxwellsEq]

This method definitely seems thourough. My only question about it is how much fatigue comes into play. After 10+ tests, depending on how long, what the content is and what levels, people could just be over it and not want to hear a difference anymore just to get it over with, lol!

This is how it has been done, by professionals trained to detect errors, in measurement laboratories for years. The evidence consistently confirms what we can and can't hear. Most of what reviewers think they hear is either an error in matching levels or is psychological bias often caused by visual cues. Sometimes it comes from the words used: CDs are shiny, vinyl is dark and soft; silver wire is bright, copper is soft and warm; silver front panels are crisp, champagne front panels are smooth etc. etc.

 

[Doodski]

This is how it has been done, by professionals trained to detect errors, in measurement laboratories for years. The evidence consistently confirms what we can and can't hear. Most of what reviewers think they hear is either an error in matching levels or is psychological bias often caused by visual cues. Sometimes it comes from the words used: CDs are shiny, vinyl is dark and soft; silver wire is bright, copper is soft and warm; silver front panels are crisp, champagne front panels are smooth etc. etc.

All very goods points and I'll add a couple too.
Titanium is brittle and harsh sounding.
Soft dome tweeters are softer/smoother sounding than metal.
and last but not least...
Eating chocolate while listening creates a chocolatey midrange.

 

[Slave IV]

This is how it has been done, by professionals trained to detect errors, in measurement laboratories for years. The evidence consistently confirms what we can and can't hear. Most of what reviewers think they hear is either an error in matching levels or is psychological bias often caused by visual cues. Sometimes it comes from the words used: CDs are shiny, vinyl is dark and soft; silver wire is bright, copper is soft and warm; silver front panels are crisp, champagne front panels are smooth etc. etc.

All very goods points and I'll add a couple too.
Titanium is brittle and harsh sounding.
Soft dome tweeters are softer/smoother sounding than metal.
and last but not least...
Eating chocolate while listening creates a chocolatey midrange.

I can relate to all of this
Thanks for all of it, really good discussion for me.

I’m usually the one to dismiss marketing nonsense and love the objectivity here. Not sure how this all explains the differences I heard in the other examples I gave and how they were able to match up without knowing what was what but I guess that will be an ongoing thing for me to look into.

 

[Doodski]

I can relate to all of this
Thanks for all of it, really good discussion for me.

I'm glad you are having a good time here @ ASR. Lots to read and look at.

 

[DonH56]

This method definitely seems thourough. My only question about it is how much fatigue comes into play. After 10+ tests, depending on how long, what the content is and what levels, people could just be over it and not want to hear a difference anymore just to get it over with, lol!
<elided>

Variety of test (source) material, friendly atmosphere, lots of breaks, and multiple sessions spread over time (in my case often weeks to months).

 

[Chrispy]

All very goods points and I'll add a couple too.
Titanium is brittle and harsh sounding.
Soft dome tweeters are softer/smoother sounding than metal.
and last but not least...
Eating chocolate while listening creates a chocolatey midrange.

LOL

 

[Slave IV]

I'm glad you are having a good time here @ ASR. Lots to read and look at.

Yes, I’ve found tons of good info here before I decided to post. Really my kind of community and I hope I can contribute in some way.

Variety of test (source) material, friendly atmosphere, lots of breaks, and multiple sessions spread over time (in my case often weeks to months).

Sounds about right. I have some tools and have begun the journey.

 

[dualazmak]

If you OP @Slave IV would be further interested, please find details of my latest multichannel multi-SP-driver multi-amplifier audio system in my post #931 on my project thread;
- The latest system setup of my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig, including updated startup/ignition sequences and shutdown sequences: as of June 26, 2024: #931

Wow! This is some incredible detail and dedication! Thanks for sharing! I haven’t gone through all of it closely yet but I noticed you are crossing over your subwoofer feed with a 12db/octave at 50hz and additionally with the subwoofer’s built in crossover at 55hz 24db/octave. I’ve done similar myself in the past with some setups but I’m curious as to what the specific intention of doing this in your case is, if any.

As for general guidance and suggestions for XO between subwoofer(s) and main woofers, my recent post here would be of your reference;

Spoiler

We have so many critical factors for crossover (XO) between subwoofer(s) (SWs) and main woofer(s) (WOs) in our audio system for better/best low frequency (Fq) sound reproduction in our own/individual listening environments including room acoustics. Of course, the final goal would be greatly dependent on your music listening personal preferences, and hence there would be no general/standard procedures and approaches, I assume.

The major factors would be;
A1. Room acoustic mode(s) including reflection, dispersion, absorption, standing waves, resonances, etc.
A2. Precision (1 ms precision) time-alignment (a kind/side of phase tuning) between SWs and WOs, at your listening position,
A3. Optimization of relative gains for SWs and WOs, should be flexibly controlled on-the-fly while listening to music,
A4. Optimal selection of XO filter type(s) (i.e. BW, LR, Bessel, etc.), XO Fq, slopes at both side, phase inverse or not, further specific EQs or not, etc.

Before starting your optimization/tuning exploration journey in this regard, you need to know/understand several features/aspects; the major points would be;
B1. "Major" low Fq sound reflective plane/wall in your acoustic environment (not always needed to be fully eliminated),
B2. Be aware of that we always have overlapped Fq zone where SWs and WOs sing together, whatever XO Fq and slopes we would use,
B3. Basic understandings on the physical configuration(s) of SWs and WOs, especially ported or sealed, difference in mass of moving parts, etc.,
B4. Difference in transient behavior (step response?) of SWs and WOs, both kick-up responses and fade-out patterns;
____Damping factor/performance of the amplifier(s) driving SWs and WOs more-or-less do "matter" for transient behaviors.

As you may well aware, many audiophile people use REW and/or similar advanced audio measurement/tuning software tools together with suitable measurement microphone(s) for the optimization. I too used wonderful REW during my early stage in my multichannel project as you can find my posts #17, #18, #20, #21, #22 on my project thread.

Because of various reasons, however, nowadays, I use REW mainly as validation and confirmation tool for my rather primitive but reliable reproducible understandable (to me) validated simpler measurement and tuning methods as shared below; If you would be interested, please read carefully these posts on my project thread;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision (time-shifted) pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507
- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics: #498

I believe the above linked my posts well cover almost all the aspects and tunings relating to above A1 through A4 and B1 through B4.

Let me emphasize that the use of rectangular-sine-tone-bust signals (8-wave, 3-wave, and even 1-wave) of various Fq and the analysis of the recorded air sound (by second independent PC) of these tone-burst signals using Adobe Audition (or Audacity) would give you really useful information on optimization of SWs and WOs. Especially the 3D (gain-Fq-time) color spectrum of Adobe Audition showing "3D sound energy distribution" is much useful (at least to me!) for observing and tuning the XO configuration for SWs and WOs (and other SP drivers). You can find typical example case in my posts #503 and #507.

If you would be seriously interested in using the test tone signal tracks I prepared and applied in these my measurements and tunings, please simply PM me writing your wish.

These posts would be also of your interest and reference;
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520

- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687

- Reproduction and listening/hearing/feeling sensations to 16 Hz (organ) sound with my DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo audio system having big-heavy active L&R sub-woofers: #782

- A nice smooth-jazz album for bass (low Fq) and higher Fq tonality check and tuning: #910, #63(remote thread)


Furthermore, I highly recommend you to establish your own consistent "reference/sampler music playlist" consists of tracks of various genres hopefully fitting well for your/our music preferences.

At least in my case, I have been using my own consistent "reference/sampler music playlist" consists of 60 tracks as I shared here #670 on my project thread, and also I have dedicated thread;
- An Attempt Sharing Reference Quality Music Playlist: at least a portion and/or whole track being analyzed by 3D color spectrum of Adobe Audition

You would please find details of my latest audio setup, well covering all of the above mentioned topics, in my post here #931 on my project thread.
- The latest system setup of my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig, including updated startup/ignition sequences and shutdown sequences: as of June 26, 2024: #931

I do hope much pleasure and success in optimization efforts in your audio system.

 

[Slave IV]

As for general guidance and suggestions for XO between subwoofer(s) and main woofers, my recent post here would be of your reference;

Spoiler

We have so many critical factors for crossover (XO) between subwoofer(s) (SWs) and main woofer(s) (WOs) in our audio system for better/best low frequency (Fq) sound reproduction in our own/individual listening environments including room acoustics. Of course, the final goal would be greatly dependent on your music listening personal preferences, and hence there would be no general/standard procedures and approaches, I assume.

The major factors would be;
A1. Room acoustic mode(s) including reflection, dispersion, absorption, standing waves, resonances, etc.
A2. Precision (1 ms precision) time-alignment (a kind/side of phase tuning) between SWs and WOs, at your listening position,
A3. Optimization of relative gains for SWs and WOs, should be flexibly controlled on-the-fly while listening to music,
A4. Optimal selection of XO filter type(s) (i.e. BW, LR, Bessel, etc.), XO Fq, slopes at both side, phase inverse or not, further specific EQs or not, etc.

Before starting your optimization/tuning exploration journey in this regard, you need to know/understand several features/aspects; the major points would be;
B1. "Major" low Fq sound reflective plane/wall in your acoustic environment (not always needed to be fully eliminated),
B2. Be aware of that we always have overlapped Fq zone where SWs and WOs sing together, whatever XO Fq and slopes we would use,
B3. Basic understandings on the physical configuration(s) of SWs and WOs, especially ported or sealed, difference in mass of moving parts, etc.,
B4. Difference in transient behavior (step response?) of SWs and WOs, both kick-up responses and fade-out patterns;
____Damping factor/performance of the amplifier(s) driving SWs and WOs more-or-less do "matter" for transient behaviors.

As you may well aware, many audiophile people use REW and/or similar advanced audio measurement/tuning software tools together with suitable measurement microphone(s) for the optimization. I too used wonderful REW during my early stage in my multichannel project as you can find my posts #17, #18, #20, #21, #22 on my project thread.

Because of various reasons, however, nowadays, I use REW mainly as validation and confirmation tool for my rather primitive but reliable reproducible understandable (to me) validated simpler measurement and tuning methods as shared below; If you would be interested, please read carefully these posts on my project thread;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision (time-shifted) pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507
- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics: #498

I believe the above linked my posts well cover almost all the aspects and tunings relating to above A1 through A4 and B1 through B4.

Let me emphasize that the use of rectangular-sine-tone-bust signals (8-wave, 3-wave, and even 1-wave) of various Fq and the analysis of the recorded air sound (by second independent PC) of these tone-burst signals using Adobe Audition (or Audacity) would give you really useful information on optimization of SWs and WOs. Especially the 3D (gain-Fq-time) color spectrum of Adobe Audition showing "3D sound energy distribution" is much useful (at least to me!) for observing and tuning the XO configuration for SWs and WOs (and other SP drivers). You can find typical example case in my posts #503 and #507.

If you would be seriously interested in using the test tone signal tracks I prepared and applied in these my measurements and tunings, please simply PM me writing your wish.

These posts would be also of your interest and reference;
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520

- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687

- Reproduction and listening/hearing/feeling sensations to 16 Hz (organ) sound with my DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo audio system having big-heavy active L&R sub-woofers: #782

- A nice smooth-jazz album for bass (low Fq) and higher Fq tonality check and tuning: #910, #63(remote thread)


Furthermore, I highly recommend you to establish your own consistent "reference/sampler music playlist" consists of tracks of various genres hopefully fitting well for your/our music preferences.

At least in my case, I have been using my own consistent "reference/sampler music playlist" consists of 60 tracks as I shared here #670 on my project thread, and also I have dedicated thread;
- An Attempt Sharing Reference Quality Music Playlist: at least a portion and/or whole track being analyzed by 3D color spectrum of Adobe Audition

You would please find details of my latest audio setup, well covering all of the above mentioned topics, in my post here #931 on my project thread.
- The latest system setup of my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig, including updated startup/ignition sequences and shutdown sequences: as of June 26, 2024: #931

I do hope much pleasure and success in optimization efforts in your audio system.

I didn’t see anything specifically addressing my question about your dual crossover setup but there is a lot there so I could have easily missed it. It’s something I will continue to look into as I am genuinely curious what the intended results and benefits are.

 

[Wolf11Man]

If you have developed a good hearing ability, you may be able to spot very small differences in sound.
One kind are some very small samples where something doesn't sound right. This may be a real fault of some component, a speaker or amp, even CD player or DAC. Something that has keept me busy for years, trying to identify and erase these quite rare annoyances. This often consolidates with longer, relaxed listening, not in concentrated, stressing listening sessions. Like when you urgently want to hear something in a test situation, because someone tells you to decide on A or B. Even worse when some more "listeners" are trying to hear what some "moderator" presents. I still remember the paper sniplets of the Linn voodoo masters under the LP something turntable. You had to hear it or showed your incompetence.

Then there are differences in sound, that can't be put in the right/ wrong or better/ worse drawer. These may be a half dB to much on a stereo set-up or a speaker that has another angle to the listening spot. Different tweeter can be such a thing, both good, but somehow not sounding identical. Capacitors fall into this category, you often can not say one is better, you just feel a very little change in sound. Don't force yourself to put such differences in the better or worse category, just accept them, like a blonde or brunette hair color. Both can be just fine. Pick your favorite or love both...

I think this can be very important: Not any difference is a fault.