This subject of digital crossovers became quite popular during last few years and many audio-people give up and go for a simplicity and painlessness of digital crossovering. There is an army of people out there who would swear by a complete transparency and non-intrusiveness of d-crossovering. Probably nowhere the subject of d-crossovering is as important as in the horn world. D-crossovering primary is so popular in the horn universe not because the crossovering itself but because the necessity to deal with delays while you do horns and there is no simpler way to introduce delays (in practically at HF) then using a digital domain. So, if people go for the D-delays then it would be very reasonable to use d-crossovers right there.

I always was a vocal and persistent opponent of D-crossovering and D-delaying. However it is virtually imposable to talk with horn people about the subject and there are many reasons why. For whatever reasons each and single audio-person is so preoccupied with the specific result that they got in thier rooms that to ask an audio-person to look at the problem wider and reflective is virtually imposable. In the end of all arguments, evidences and opinions a person usually say’s something like this say: “Come on, Romy, when I put my Behringer DCX2496 into my system and replaced my passive speaker-level, passive line-level, active-passive, active-feedback, active-subtraction and so on …. then I got a way better results”. Well, I am glad you did but:

1) There are no references how good your analog filters were implemented before – whatever I’ve seen so far was bad up to the point of being atrocious. Put in this way: there are no first-rate commercial active crossovers – no mater how expansive they would be.
2) 99% people who are advocating D-crossovers and D-delays campaigned this view using very appropriately, or at least non-seriously bult playback systems. (PA level amplification, grossly compromised horns installations and so on…) and very mistaken/primitive evaluation methods to gauge the result.
3) The domination majority of D-crossovers and D-delays supporters are not qualified properly evaluate a playback result. To bult up system using the objectives like: “When I played my jazz the my legs were dancing” or “Look, the Patricia Barbers sounded on my system so good than that my cactuses begun to blossom” is not the level where my believes operate. I know that I would endure an accusation of being arrogantly-foolish but I insist that absolutely dominating majorities of people in audio are completely dead regarding the understanding of what the deal in audio and what they do in audio.  Therefore the subjective assessments of the D-crossovering from a prospective of a foolishly-exuberant and essentially-dead audiophilic community is completely irrelevant to me. There are extremely small number of individuals, perhaps a couple hundred people around the world, who might intelligently, sensibly, judiciously, knowledgably and objectively assess a performance of playback but the rest is an ballast sound-evaluation-wise
4) Despite all bold promises from many individuals “Romy, my playback sound fantastic with D-crossovers” I NEVER experienced any interesting sound when I actually heard those systems. Nope, the D-crossovers were not a sourse of the problems in there but their entire playbacks had so much OTHER PROBLEMS and badly implemented compromises that there was no interest to even think about cons or pros of D-crossovering in context of those inhalations.

I have to admit that I would LOVE to see a digital solution to work well and if I find that it is the case then I would be gratefully embracing it. Unfortunately it never happened so far. My experience with this started a couple years ago when I discover an absolutely phenomenal mid-bass driver to bult 40-50H mid-bass horn. However, I needed a low-pass right after it does 1-2 octaves and to introduce 8.5 feet delay. Certainly I would not use a high-order analog filter and to introduce the delay for wide-bandwidth HF would be a sonically suicidal. So, I began to explore the D-solutions.

Whatever I tried did not work and I am not talking about crossovering but juts AD-DA conversion: an introduction of ADC-DAC element into a signal path instantaneously screwed up sound. The very minute deviation between the shadows of the tomes became less distinctive, the inflection of voices and pitches became less characteristic, the musical accents and enunciations became muted and more neutralized, music become more genetic and less idiosyncratic, more washed-out. In other words the result defeat the purposed that I was pursuing with my audio objectives. Put in this way if I conduct an orchestra that would play this way then I would fire the musicians, so why should I go in there with my playback? All that I was describing happened with all 5-6 processors that I trued. All of this affectively removed among the available options the idea of D-delaying. The only one solution that I fine interesting, was to have a pair of DACs after my CD transport connected in parallel and to bult-in a D-delay into one of them (his would not introduce any problems). I would certainly go there but I do have an analog setup that I play quite aggressively. What should I do with analog? So, I ended up with a fiasco to find a good solution for delay in reference to my mid-bass horn and the horn building ceremony never was initiated. The “phenomenal upper bass driver” is sitting in my storage unit the 21-century civilization find a way to make delays that work :-) or until I will be able to do it naturally by appropriate positioning of the horn

Now the funny part: the D-crossovers. As that time I got the Behringer DCX2496 and decided to experimented with it crossovering possibility. As soon I activated the crossovering the result was even more catastrophic then I described in previous paragraph. It was like someone inject fog into music, bleached it out, compress it, along with countless other problems. All those problems were of the magnitude that there is no needs to talk about anything further but juts trash the crossover and never touch it. Later on, I borrowed form my local pro shop all-possible the most expansive crossovers and the result was identical: as soon the curve was introduce the sound when down. The very same was experiencing when I played with countless CD recorders – as soon I touched that fade option or anything similar DSP-based, then sound went immediately into a toilet. My experience with very many preamps that had digital remote controls was pretty much identical, so I decided that it was not the direction to go.

Then I begin to approach a number of the individuals, the world-class digital designers who designed whatever you today considered the best in the digital domain. They expanded me (and this was a very extraordinary occasion when ALL OF THEM were agree on this subject UNANIMOUSLY) that DSP, no mater how good it implement, even for with no price restrictions, is a fundamentally sound screwing concept. As soon a curve introduced it done by removing bytes at the bottom of the curve and as a result it introduce problems of loosing resolution and dynamics at lower levels. Country to what thy explained me I clearly hears a severe problem not only near the slopes but even at band-pass. Nevertheless, they all (all of them!!!) told me:  “It is not about to bult a properly sounding D-crossover but to make it that a given user with his/her reference points would not be able to become aware of a sonic degradation”. I would not bring the names of those people to back up my story because some of them today … are manufacturing digital crossovers and some of them are about to release D-crossovered loudspeakers. However, I might assure you that they are the most respected and the most competent names in today digital sound possessing. I have to note that I was very serious about the project and I was ready to pay $20K-$30K for a digital delay-crossover that would not introduce any noticeable by me deterioration into a signal path. When the designers learned about my objectives they suggested me that I would not find a solution with their digital world. So, I abandoned this direction.

Now, some further observations on the subject and why I consider that generally people who dream about the beauty of DSP processing with horns looks at fundamentally wrong direction and think kind of wrongly.

First off all: why do we, the horn people, ever need a digital crossover? (I am not talking about the delays now). People who know me know that I religiously multi-channel guy and that I believe that so-called full-rage channels is a celebration of a wishful thinking and sonic idiocy over really and actual results. So, the only know to me way to make a playback sound acceptable in context of multi-channel is to design the channels that have no-electrical low-pass filters (use acoustical filters, throat EQ or a nature driver decay) and use no higher then first-order electrical high-passes. Let me to repeat: 6dB per octave eclectically – nothing else!!! So, we have one single cap that should be used in the entire channel path. So use it! Place it wherever you wish, at speaker level works very fine, way-way-way better then any imaginable D-crossover. I have to report the even at speaker level a capacitors, although it might have some different “sounds” and tonal influences, but the NEVER destroys the IMPORTANT attributes of sound responsible for musicality. The speaker level capacitor affects audiophilism but it dose not screw up music.  If you wish to go further then instead of a speaker level cap place instead of the across-stages coupling cup if your amp use them and the design permits. Alternately you might run this single serial cap against a high-impedance of your line level stages, consequentially reducing it’s value to pF or nF level and using some high-quality caps (air, teflon, vacuum and so in). The little tonal deviations that they might have are completely irrelevant at speaker level and of course it you place a superb quality ultra-low values cap at line-level then you have absolutely no-deteriorating result. So, why in the hell we need a digital crossover?

Some people would disagree with me. Those people, primary the GOTO, ALE, TOA and BMS users, would suggest that a proper horn-loaded reproduction is imposable within a channel operating with a none-limited range and therefore, by used a multiple very narrow bandwidth channels, we have to use a higher-order crossovers; so, considering that the 2-4 order crossovers are imposable to use at analog domain we have to go digital.  I would very much argue this point of view. The RCA’s Radiotron clearly suggests in 20.4 that limitation of frequency ranges relates to a desirable efficiency and my experiments do confirm this observation. With a park of existing drivers is it perfectly possible to accomplish 110dB of sensitively via 4 channels. As soon you go higher then approximately 110dB-11.5dB then you do need to introduce one more channel but do we REALLY need to go there? The GOTO and ALE would be happy to sell you 10 differents drivers for each single octave each, ironically archiving no more then 110dB sensitively but forcing the system users to buy more drivers (it never hurts, doesn’t it), facing the person to face a crossovering nightmare and the dilemmas of the horns alignment/positioning. So, do the artificial believe in a horn systems with 453 channels has any practical rational? I do not think so. Therefore, I do not see any arguments for building 5-6 channels systems. It would be ironic to mention that the word suppliers of GOTO and ALE drivers are… “accidentally” the distributors of Behringer and BSS digital crossovers.

(I would accept, and agree with some arguments about the excessive amount of LF channels: like low-LF, mid-LF and upper-LF channels but with the LF is totally different story and here is where the higher order crossovers might be utilized. However, bass channels is totally different conversation)

So, were I am staying? Certainly I propose that the digital crossovers should not have a place in horn installation if the system is designed rationally, knowledgeably and with a respect to the Result. They were my words, my experience and my thinking. Your mileage might and most probably will wary. However, so far I did not hear any result that would encourage me to review my position and all installations that I heard (not only horns) that used D-crossovering were at sub realy serious level.

PS: I have to make this disclaimer because most of Stereophiles are sick on their heads and would consider this article as a personal attack on their damn playbacks. The Morons: I hope you understand that this writing has a pure conceptual and educational purpose and it does not call you to take a hammer and crash your d-crossover. Do, think and believe in whatever makes you happy and if you do used a digital crossover with horns then it is YOU who benefited with the result. Perhaps you "deserve" it….

PSS: For whoever who does use the D-crossovers:

Sometimes ago I argued the point with a French guy: Jean-Michel Le Cleac'h somewhere on the Web. Unfortunately that conversation went to nowhere because that site where the conversation took place was moderated by some kind of idiot who in order to kiss that French guy's ass was deleting the posts that disagree with Jean-Michel’s view. Anyhow, if you do use the Behringer DCX2496 or BSS FDS388 them you most likely experiences the very crappy sound from your D-unit (it is what I always heard from YOUR systems). Jean-Michel offered an explanation for it. I did not try it (when I used the D-filers I was resolving the unity-gain via different means) but his explanation sound quite reasonable and you might consider it.

Here are the Jean-Michel comments:

“When I first replaced my previous active Kaneda crossover, which one used discrete stages and was excellent with the BSS FDS388 I was really disappointed. The sound was dirty, noisy, harsh... then I measured how many volts RMS I had on the input: 80mV. This meant that I had +37dB digital noise added. Then I put step down transformers and attenuators on every outputs and feed the inputs at high level (attention to clipping!) this was the best crossover I ever possessed. Please notice that I have also transformers at the input and that the crossover input lines and output lines are balanced.”

“A device like a digital crossover when used with analog inputs and analog outputs possess a gain of 1 (well in fact there is also a digital attenuator included generally but if you use it you loose also some bits and some precision in the output signal) I don't know what is the sensitivity of your amplifier but most surely for the max acoustic level in your listening room the input signal of your amplifier is probably less than 1V RMS. This means that you have also 1 volt RMS at the inputs. To obtain the best accuracy of a digital crossover as the Behringer DCX2496 you have to feed it at the inputs with a max signal of 10 VRMS. If you feed it with a 1 V RMS max you loose 20dB on the ADC and the DAC. Our experience is that this bad use of the ADC and DAC overall dynamic is the main explanation of the bad sound that few reported with digital crossover. To use a digital crossover you have to use the correct input level and this often means the use of a larger preamplifier gain. In consequence you have also to put an attenuator at the output in order that the power amplifier input doesn't saturate.”

Nice posts Romy and Jan, agree almost 100%. I've heard the Behringer in an originally crummy system and it butchered things completely. The input levels were low, which as you point out, won't have helped at all.

On a vaguely related note: I dunno if you realise this, but conventional (ie excluding horn or large area ESL) speaker systems, even if measuring flat amplitude, introduce 90 degrees of phase shift; at 100Hz, this is about 3 ft worth of delay, and it gets worse the lower the frequency.

From this standpoint, I never understood why users of multi-tower systems (eg the big Alons, IRS V style etc) always seem to have the woofer towers positioned behind the mains. Equally, "time-alignment" of the LF with the mains is a misnomer.

Least compromised approach is proper horn loading to get the delay vs frequency (which is what phase is really) flat and the get the mastertapes and play them back on a machine with 2 heads to get your delay :-)

Although all of it has no relation to digital crossovers but still:

cv wrote:

On a vaguely related note: I dunno if you realise this, but conventional (ie excluding horn or large area ESL) speaker systems, even if measuring flat amplitude, introduce 90 degrees of phase shift; at 100Hz, this is about 3 ft worth of delay, and it gets worse the lower the frequency.

Sure they do, all of them and there are many reasons why and how. However, the fear of “phase shift” should be properly understood. A system should be phase-coherent initially, design-wise…. but THEN a system MIGHT introduces some beneficial phase anomalies. However they should be intentional, well understood and evaluated. By injecting into room some “strategic” LF phase irregularity it is possible to get some very “interesting” results that might very effectively overwrite/mask-out some other probolems of sound reproductive environment.

For an experiment walk around a not-tightly seated large symphonic orchestra while the’re playing and see how your inner-you reacts to the LF delays.  For instance if the orchestra positioned with bass strings and with tubas at upper right (the typical contemporary orchestra positioning) and if you’re 20 feet in front of the orchestra on the left then you have a approximately 50 feet between the first violins and bass section. If disregard the gross luck of tonal balance in this case you might start to walk back and forth, right and left and you might find out then that it is not about a “correctness” of the delays but rather about the audible benefits. Even considering the phase randomanization in the concert hall you still might see that the delays might be a quite powerful EXPRESSIVE TOOL. Why do you think that managing delays within our playbacks we are in very different position then conductors who manage the delays of the orchestral groups? Good conductors recognize the idiosyncratic acoustic conditions of the specific performance halls and tune thier orchestras accordingly. We should do the very same with our rooms and with our dedicated to the specific frequencies channels.

cv wrote:

From this standpoint, I never understood why users of multi-tower systems (eg the big Alons, IRS V style etc) always seem to have the woofer towers positioned behind the mains. Equally, "time-alignment" of the LF with the mains is a misnomer.

Come on Chris! Those systems are a celebration of consumer deception over a common scene. Those multi-tower systems system never bult to accomplish any sound Result but rather to make a product presentable, large, size-tangible and be able to convert the presumptuousness of a products into money. Theoretically it should be wonderful: a company manufactures a LF-restricted loudspeaker and they provide as specially designed LF module that should be used with it. Sounds all kosher, does isn’t it? In the reality: none of those “specially designed LF modules” sound even remotely interesting or even marginally acceptable. They all made to bypass nothing further then a primitive reference point of industry’s quality control - the audio reviewers layer, and there were no further more noble objectives in the heads of their designers. Listen to at all those Martin Logan Statements, Pipedreams, Genesis, Alón Exotica Grands, Gryphons and the rest of them with passive dedicated LF sections. Why they all sound do predictably horrible? Evan if the companies go for looked-like-no-compromise pretensions then they still allow own stupidity and own greed to overwrite OUR listening benefits. Look for instance at the David Wilson XS woofers. With 800 pounds of a wonderful built inside of 45cu-feet of those sarcophagus and the price tag if I remember correctly $29000 each… How come that they still introduce a half-ass solution? Why in the name of God, using a pair of amassing Aurasound 1808 drivers per channel, they went of ported enclosures? David told me: “We have +20dB output ay 20Hz compare to a sealed enclosure”. So, what? Does he think that anybody drive them with 2A3 SET amplifiers? Or a person who have spent money for them and was able to accommodate a pair of them in his listening room would ever care? A pair of 1808 have 101dB sensitivity and in this enclosure they might be an spectacular LF section but… the “industry” again converted any would-be-good-intention into a “popcorn blowing machine”….

cv wrote:

Least compromised approach is proper horn loading to get the delay vs frequency (which is what phase is really) flat and the get the mastertapes and play them back on a machine with 2 heads to get your delay :-)

I actually considered the approach: to try a very good tape delay unit from 70s. If I would need to delay one channel I would probably go there (I do like how tape sounds) but I would be hesitant to do it with full range.

There are no set of laws in there - you have to play with it and then you might ketch something. Certainly for a proper bass reproduction (or generally music reproduction) you need somewhere around 1 sec decay in your room. You might have a large room or to get a longer reverberation time via different means. However, I detected that when I VERY SLIGHTLY delay the arriving time at the crossover point of my woofer towers then I get very “interesting” result. I have in there now a second order at line-level, a perfect Bessel curve  (minus 79 degree). The woofers are ACOUSTICALLY synchronized regarding the arrival. Nevertheless, I always run them 1-1.5 mS later then horns. Interestingly that the very same behavior I observe with 1, 3 4 orders and with other LF modules that I used. I do not know if it is the behavior of my room of a general behavior as I never play with other system to this level of precision but I feel that there is something in it and I more inclined to accept it as an universal pattern.

Rgs,
The Cat

Hi,

Romy the Cat wrote:

This subject of digital crossovers became quite popular during last few years and many audio-people give up and go for a simplicity and painlessness of digital crossovering.

Most readily available digital X-overs come from a Pro-Audio background. Even if we ingnore the many inherent items that will compromise the performance of such devices, there is a crucial problem linking them into a home audio system, namely the level issue.

Let us assume we have a "state of the art" digital Pro-Audio crossover with perfect DSP and good quality converters allowing per output a dynamic range of 114db (which is not bad if not the best achievable) or a real 19-Bit resolution. The full scale level of that X-Over is for arguments sake at +22dbu or 9.8V RMS. Let us say we use a Driver of 105db/1W/1m and we connect between X-Over and speaker a Lamm ML-2 with a nominal input sensitivity of 0.775V (or 0dbu) for 18W Output.

Now, the resulting would have AT FULL POWER from the ML-2 a SPL of 118db @ 1m. The noisefloor and resolution limit of our digital X-Over would be at -114db re +22dbu but only -92db re 0dbu, the point where the amplifier is likely to clip. Now if we assume that normal listening levels for a big orchestra at FFFF is something like 105db peak @ 1m with this system or 13db lower than "full tilt" of the Amplifier we now have the system "resolution" (not neccsearily the noise floor though) at -79db with reference to the highest peaks during use.

In other words the music begins to turn to crap at levels around 80db below max....

In other words still, we have just succeeded in making our system a digital system with around 13 Bit actual resolution.

Now, the solution. Insert ONE AD-1 after EVERY output of the digital X-Over. Adjust the level on the AD-1 such that at the maximum levels normally listened to you retain around 6db headroom on the peak reading meters build into any such Pro Audio Digital Crossover (PADXO from here on). You may find that the PADXO output will clip before your amplifiers but if you never listen that loud, what the heck?

So, our system is now fitted with AD-1 between PADXO and ML-2. With an SPL of 111db/1m the PADXO reports "over". So, the full dynamic range of the source is retained and any "artifacts" from the X-Overs internal processing etc are suddenly for any given listening level around 33db lower than before.

Of course, most PADXO's (especially the Behringer DCX-2496 - see footnote 1) have analogue stages and powersupplies that are open to severe questioning and in fact suitable upgrades can increase dynamic range and improve the subjective soundquality to a very large degree.

EVEN THEN the only 32-Bit DSP system in most PADXO's may present additional problems that may become audible, as one would really need 64-Bit or larger floating point number crunching ability to ensure complex filter algorythms can be implemented without running out of resolution. So, the more you EQ and the steeper the Crossover curves the more problems you generat for the DSP.

So, affordable PADXO's are a pretty good tool and used well can work very well, but they will (not yet) match a really well implemented line level passive X-Over/EQ system. That said, if you implement the PADXO well into the system and you have done the neccesary mods the difference becomes somewhat notional and for most semi-demoronised audio morons perfectly livable.

Ciao T

Footnote 1 - The analogue stages of the Behringer DCX2496 and DEQ2496 are a textbook example how NOT to do it.

Their old DEQ8024 Digital Equaliser has excellent designs of analogue stages only compromised by parts quality, there a simple Op-Amp buffer with a "electronic transfomrer" servo type circuit was used in the input stage, placing just 1 Op-Amp in unity gain mode is series with each polarity for a balanced signal and for an unbalanced signal there would in effect be another Op-Amp as inveter on the non-driven input. The output stage had again 1 Op-Amp per polarity with another electronic servo circuit, there was a single Coupling Cap per polarity in the Input and equally in the output. Muting was via relais. About as minimal a stage count as you can get. The AD & DA chips where a little iffy, sadly, especially the DA.

Now their new stuff input circuitry first has a pair of coupling electrolytics to one Op-Amp to turn the signal from a balanced one to SE and to attenuate it, then another coupling cap to another Op-Ampto re-amplify this signal and yet another Op-Amp to regenerate a balanced signal to drive the ADC.

But the "Piece de resistance" is the Output. Here they take the nice balanced output signal from our DAC and turn it SE with an Op-Amp. They then couple the signal via an electrolytic cap to another Op-Amp and implement muting via muting transistors (YUCK) between the two. Following these two Op-Amp's are TWO MORE, to make a balanced output from the single ended signal. Just for good measure all resistor values in the circuit are pretty low, so the distortion from the Class AB output stage of teh Op-Amp's used is maximised.

If one wanted to, the DCX could be modified to accept SE inputs and outputs with a minimal number of highest possible quality Op-Amp's (OPA627 or similar on input, LM6171/72 r similar on output) or even without any Op-Amp's if we employ suitable transformers to replace all the solid state circuitry.... Either way result would be much improved and not require silly levels of attenuation to match HiFi gear as well.

slowmotion wrote:

Good news if the analogue stages of the DCX2496 is so bad. So the improvement of a modified unit should be worth the ekstra work, and remove some of the x-overs negative influence on the sound. Which is always good to know beforehand. There's always compromises, of course, but with an all horn setup the plus outweighs the minuses, IMHO. Have you looked into using computers for x-over duty?

It is not about the compromises but about the fundamental damage that DSP inflicts to sound. People who use digital crossovers, because sound was already screed by the crossovers, juts do not know how their drivers perfume in reality. I’m not even talking that the people who use d-crossovers have no touch with driver’s needs as the drivers could NOT be use in their environment without being digitally crossed. How do you know how your drivers really sound at the given ranges? How could you say that your digital crossover “does well” if you have no ways to use the drivers without digitalization?

Anyhow, I did not look for using computers for x-over duty. It is not about the computer vs. dedicated crossover box, of better A/D, D/A, or a better output stage. That all is hard to accomplish but it is accomplishable. The problem begin what you inject a digital slop into auditable range, loosing resolution and injecting sonic fog into each decibel of lowered volume.

There is an unavoidable rule: digital can’t filter, analog can’t delay (practically at LF). It is like humans do not eat sand and do not drink ocean water. Sure, it is possible to do it and call it “compromise” but what the purpose. There is HUGE amount of benefits (not only sonic!!!) with analog filters. I do not call using digital filters as compromised but rather a victory of wishful thinking over a sense Realty.

Try also this test:

http://www.goodsoundclub.com/TreeItem.aspx?PostID=2650

Rgs,
Romy the Cat

slowmotion wrote:

Romy, there are more ways to the top of the mountain... <BR>And we already know that your way isn't my way... <BR><BR>You bring up many very valid points, but still I will continue to use my digital crossover as a very useful tool. But I am also experimenting with other solutions...

Pay attention that I am not bitchy about digital delay (although I do not know any transparent enough) but only about the digital crossover. There is another “philosophical” problem with digital crossover. You see, to design crossovers for horns is a quite sophisticated art. To design crossovers for box speaker is complex but in there we correct the drivers’ and enclosures’ problems. With horns, we do not correct problems (if proper driver and horns were used) but rather identify the most desirable operation conditions and let the drivers/horns to be as they are in those conditions. It is very tiny and not distinctive difference but it does exist and it affects sound fairly dramatically.

So my point is that when we “manage” a horn channel via a line level digital crossover we do not have the feel of what the horn/driver does. Use a digital crossover is like patting Cat by means of remote condoled mechanical arms: you can make the Cat to purr but you will not get the reaction of her body… When I use D-crossovers I have different reaction then with analog filters, and I am not talking about “quality of sound”. Let me explain. I have a RTA and I’m looking at the specific slope of a specific driver/horn.  I hear a certain sound from this driver/horn. Then I decided for instance to move the slope for a couple hundreds Hertz up or down. I do so via analog filter and get the well predicable and expected sonic gratification. With digital crossover, if I mover the slope at the very same amount of Hertz the sonic delta is different. I do not know how to explain the difference but I do feel that it does exist. I feel that with analog filters the difference is something that I would call “predictable”. There is a direct sensible relation between the “numbers in Hertz” and the “sound”. With digital crossover it is also exist but the relation between Hertzs and sound is less linear and way less predicable, or I would say expectable…

Rgs,
Romy The caT

Sorry, disregard, I found the threads where you previously discussed this subject.