Speakers 10 years old or older that can compete with todays best,

@mikelavigne wrote:

my room is epic, my set-up and room tuning is epic, any sort of dsp would be wasted and regressive in my particular room and signal path. no matter the acronym.

Impressive looking setup and listening room indeed.

Not to unnecessarily stir up the "why no DSP?"-question that appears to have been aimed your way already as an option with your system, but have you - in the analogue domain with an electronic crossover - experimented with an outboard active configuration at some point? I’d also add that a DSP can act as a digital crossover only (wholly replacing a passive ditto), sans room correction, but of course that still involves the "intervention" of a conversion to and from a digital processing part.

Though you have no doubt come to a conclusion on this matter, from my chair - and with a digital source only - the use of a DSP acting as a digital crossover is thinking about the passive counterpart it replaces, and which of the two is the lesser evil. Assessing a DSP section as such comes in conjunction with the important negation of the passive crossover to offer a direct driver-to-dedicated-amp-channel connection which, in the different converted from passive to active setups I’ve heard (that is, maintaining the same main speakers), has always led to an advantageous outcome - by a comfortable mile even.

It just seems to me that many regards the insertion of a DSP (and mostly assuming it’s acting as a room correction device exclusively) without considering that it can replace a passive crossover as a digital ditto, with all that entails wrt. driver control and overall filter implementation and the elaborate settings potentially involved here. Thinking that a DSP is mainly an add-on to an existing passive setup as a room correction means, while being perhaps its primary function as they’re mostly implemented, is really only seeing it for a part of what it can do, while arguably missing out on the most important one.

@mikelavigne wrote:

no, i have not done dsp in my 2 channel room. years ago i decided instead to fix the room; building a room without limits. then tune it to work with ultimate speakers. which over the last 20 years i have done.

For this context I’m inquiring merely on the use of DSP/electronic crossover as an approach replacing a passive ditto for outboard active configuration, and not - as an outset - with anything that involves digital room correction. What you’ve done acoustically is extremely dedicated and thoroughly executed; I can only assume you’ve achieved stellar results here in conjunction with your chosen hardware/gear and overall implementation.

when you write about dsp, replacing passive crossovers, i don’t think you imagine passive crossovers that are inside the top level speakers. what that looks like, or sounds like ...

First of all, any crossover option is a potential, but irrespective of the quality of the parts a passive crossover will always be a bottleneck between the amp and speaker (i.e.: impacting an amp’s ability to control the drivers) that prevents either to be nearer their fuller performance envelope - the more so the more complex the crossover at hand, with potentially severe impedance dips and steep phase angles which seem to be more prevalent among "high-end" segment speakers, and that therefore have a tendency to require very sturdy, more or less load indifferent (and very costly) amps to perform their best.

... and when you write about driving each separate driver with it’s own amp and dsp crossover, you forget what that means in terms of choices of amplification.my darTZeel 468 mono blocks are crazy spendy and the best amps i have heard......how is that going to fit (physically and $$$) into active crossovers for each driver? the answer is that is does not fit at all. i would have to settle for less capable amplification. a compromise.

I’m not forgetting anything here, because outboard active config. doesn’t require of you to compromise with regard to amp choice. You can choose whatever amps you like this way seeing they don’t need to fit inside the speakers, however you do need more of them (as always with active config.) to feed each driver section with its separate amp channel. Remember, active config. is defined by the filtration part taking place prior to amplification on signal level, and not on the output side of the amp taking the full power as a passive approach.

Not seeing into a passive crossover, not least a complex one will be making more effective use of the power at hand, why less power is needed for the same overall SPL actively. Moreso, and importantly, the actual sonic potential of a given amp will also see an uptick being presented to a much easier load actively, so here as well you can get by with less - should you so choose. It worth noticing also the power independency between the bandwidth limited amps; the bass amp could be blasting along, and it would mean zilch to the other amps feeding the remaining driver sections. To boot: the mids/tweeter amps would be rid of any LF signals, meaning even easier load and better sound. My advice though would be to use what’s essentially the same amps top to bottom, possibly power differentiated, to maintain coherency as best as possible.

reality is that dsp does make a great deal of sense doing particular things. fixing rooms, powering more modestly priced gear, enabling DIY’s to build interesting projects. integrating subwoofers. doing multichannel such as Dolby Atmos.

It can do that for sure, but that’s still selling active short; it’s sad more audiophiles aren’t aware of the potential of active as an outboard (or, for that sake, bundled) and all-out solution, instead seeing it being met with conjecture, dogmatism even or what’s otherwise an ill-informed stance. Certainly here you could ask yourself what imparts the bigger obstacle: a passive crossover on the output side of the amps, or an active ditto feeding the signal inputs of dedicated amp channels looking directly into each of their drivers sections, conversion steps be damned. Only actual experimentation will make one the wiser.

i have a never opened box unused XILICA XP-2040 upstairs in my storage attic that i bought 3 years ago intending to use it to integrate 3 subwoofers into my Home Theater. i get what dsp can do. turned out my 3 Funk Audio 18.0 subs came with their own dsp engines, so never needed the XILICA. so i’m not anti dsp.

Xilica make very good DSP’s (it’s what I use myself).

*******

Look, I’m in no position nor do I intend to impose on you anything. The setup and room you have looks to be a true all-out approach years in the making, and one that’s rarely seen. And yet I felt slightly provoked by your all-analogue stance that I thought it interesting to tempt an active approach via DSP - only because I’ve seen it trump most any passive variant (i.e.: one and the same speakers converted from passive to active) that I’ve heard.

@mijostyn wrote:

And that sirs is the trap. What a system and room looks like and what it sounds like are two entirely different issues.

I only commented that it looked impressive, but I see no issue in assuming it sounds great.

@daveyf wrote:

@mikelavigne You are a highly experienced a’phile, why would someone believe that they could improve your system, without a) first listening to it and then b) believe that their knowledge would trump yours??...I don’t really understand this attitude from several members who have posted upstream. To these very same members, i would ask this question: Is it possible that in fact someone like Mike L could have a superior sounding system to yours...and not only that, have more experience/knowledge than you?

Speaking for myself and my replies to @mikelavigne I anticipated an above-like response, and it’s not without merit. Maybe I even hoped such a reply would be leveled at me/us so that I could better explain at least myself and not come across as an arrogant know-it-better, but rather that my intention was to challenge a single aspect of which I find myself to have some experience here that Mike, it appears, does not - at least not extensively with his 2-channel setup.

Yes, I would expect most everyone to find Mike’s system to be superior sounding to my own setup, and yes it wouldn’t surprise me if Mike - in a range a areas - has somewhat more knowledge and experience than I. Hopefully I made it clear that I can only assume it’s a truly great sounding system of his, and that the implementation in every regard, from all that I can assess, has been thoroughly considered and executed. The effort and time put into it all is certainly awe inspiring.

That being said it’s also problematic to blindly expect an individual with a highly sophisticated and expensive system like Mike’s is above criticism or suggestive acts of any kind. Having not listened to his setup implies both the fact that I can’t judge its sonic merits, just as well that I cannot take it for granted it’s a sound that - despite the money, time, research, effort and dedication that went into this - would blow me away in every single aspect. I’ve heard my share of über-expensive setups in homes that left me quite unimpressed; systems that sounded disjointed, uneven, stale, bloated, over-damped (more than under-damped, actually), overly detail focused, malnourished, etc. Very few of them sounded really natural to my ears, and it just goes to show that every setup, regardless of price, is a potential only, but also that personal preference varies. Preference I can deal with, but a badly implemented system is just a waste.

Nothing to me implies that Mike’s system is badly implemented, on the contrary. I merely suggested an approach, of which he apparently has no experience to speak of (and who among us has experience in every facet of audio reproduction approaches?), that could potentially lift the sonics to even higher levels of quality. From my chair it would be worth pursuing if this particular area (i.e.: outboard active configuration) is a stone unturned - not least with the ambition at play here.

@daveyf wrote:

One other thing i have also learned in this hobby is this: it is just as easy to go backwards in regards to SQ when you implement something into your system, as it is to go forwards...and many times, even easier!

Interestingly, the dealer I visited most recently utilizes a Linn turntable with the latest thinking by Linn on the interface between the table and the upstream phono stage. Their phono stage is now built into the table and called the Urika 2. This phono stage takes the analog signal and transforms it into the digital realm to send it upstream. Essentially, one listens to a digital signal when now playing the top flite LP12 Klimax model with Urika 2! On paper, this looks impressive, because not only is the signal taken immediately from the tonearm via a very short lead to the phono stage, but also the potential for loss of signal is now limited upstream, due to the digital conversion/aspect. Unfortunately, in real life, what I always hear when i hear this set up is the following...1) a severe lack of depth portrayal 2) a sheen that can only be considered as a digital artifact that pervades across the whole frequency spectrum and 3) a certain timbral aspect to the high frequencies that upon first listen is impressive, but actually wears on one as time progresses. My point here is that while Linn ( a well respected company within the a’phile community) believe that their way forward is superior to what has been done in the past, in my personal opinion, they have actually gone backwards. Is DSP the answer? Possibly in some systems, but I would say that an ’analog’ solution would be preferred firstly, if at all possible. At least to my ears, and IMHO.

What’s important here is to pay attention to which degree a specific solution - in this case DSP - is recommended and sought implemented from actual experience. In my case I’m speaking of the use of DSP strictly in the context as a digital crossover replacing a passive ditto for active configuration - with a digital source only. This has several implications not least of which is the removal of the passive crossover between the amp and speakers, with all that implies.

Regarding your Linn turntable example, I’ve always thought it defeats the purpose to digitize an analogue signal at the source as described. I don’t have the experience to back this up the way you do, but at least my outset could say to run in tandem with your perceived findings. In what way a DSP acting as a digital crossover later in the chain will impact the sound of an analogue source compared to going analogue all through with a passive crossover instead, I couldn’t say, but you’d have to hold this up against having direct amp-driver control with dedicated amp channels feeding each L/R driver section - again, with all that implies. This is not trivial, nor is outboard active configuration tried out by many to get a bearing on the implications of this.

My point here is that using DSP must be seen in its context and how it’s implemented. As I said, I use a digital source only, so there’s that. I don’t yet use DSP for digital room correction, but only as a digital crossover, actively, and as such that’s my recommendation of it. I’m not against using DSP for room correction, but I do find it needs to be done sparingly so not to lend any noticeable "processing imprinting." Down the road I’ll be using DRC for sure, both in the amplitude and time domain.

@mijostyn --

I use a digital source only, my DAC/preamp sports a lossless digital volume control (i.e.: no effective loss of bits at lower volumes; it's also the preferred volume control option over JRiver MC31's internal digital ditto), and a digital crossover for active config - I'm not a stranger to digital, I'd say. Implementing the digital XO-settings has been done with the aid of measurements and countless hours of listening tests, so in a sense this has been done in the analogue domain, certainly without DRC, and that's how I prefer it as of now. There's hardly an automation to this approach that can bypass a range of manual factors in setting the filter values (i.e.: by ear and experimentation), and these would come in handy with an eventual correction in both the amplitude and time domain, say, with FIR-filters. As I've said already, this will come down the road. 

@mijostyn wrote:

There is no such thing as a "lossless" digital volume control. Some units may revert to an analog volume control. But, even the best digital volume controls lose bits as the volume goes down. It is just that the really fast processors do not lose enough bits to affect sound quality. 

You're right, there isn't, but I explained what I meant by the "lossless" part in parenthesis, while forgetting the quotation marks. Whether or not the volume control of my DAC/preamp sports a 64-bit floating point (I don't know its specs in this regard, other than it's digital through and through), I can't say, but it trumps the 64-bit floating point VC of my JRiver MC31 playback software (with DAC volume at what equates to 0dB), and that's all I need to know. And bear in mind: my speaker system is ~100 to 111dB sensitive, actively configured, so oftentimes I'm using the lower range of the volume control, which only tests its resolving capabilities here.  

Down the road? I have been tuning my system one Hz at a time for 25 years. Let's say I measure my system and see a 3 dB dip in the left channel at a specific frequency say 358 Hz. I can select 358 Hz and increase it 3 dB. I can also adjust the Q of the filter matching what I see on the measurement exactly. 

That's what I'm doing already and have for a few years now, but what I've learned from measurements and the corresponding, manual frequency corrections is that a frequency peak of, say, 5dB's shouldn't be suppressed with -5dB's, but rather less. Indeed, less is more here, and it's surprising how smaller corrections can have an effect - indeed the proper effect. We measured a peak over my EV horns at just over 13kHz with the left and right horn peak measuring exactly the same down to the single Hertz, and applied a mild suppression here (i.e.: less than the peak indicated) for the best sonic outcome. Decibel increments with the Xilica DSP are adjusted at 0,25dB's, although 0,10 would've been nice. Delay in 0.010ms increments. Single Hz ditto. 

@mijostyn wrote:

And just how are you correcting amplitude 1 Hz at a time? It is not down the road for me. You can do it in an automated fashion or manually including programing delays. I start with automated then fine tune manually. I find it best to program for flat then overlay my own preference target curves which were constructed by ear. 

Depends on what's being addressed. Are we speaking notch placement or PEQ? Notches in the HF-region are located precisely with nearfield measurements, whereas PEQ's can be more of an assessment by ear from the listening position (in addition to measurements), starting out "overshooting" in larger Hz-steps (and gain ditto) to get an overall bearing, and then fine tuning in ever smaller increments and Q-width variations.  

Efficiency is nice if you want to use small amplifiers. Personally, I do not care about it. I prefer to look at the type of loudspeaker.

It's a common misconception I find think to exclusively link up high efficiency with small amps as the preferable scenario. High eff. speakers + high power amps can be great solution as well - why limit yourself to one approach, and from what, experience? I too look a the type of speakers, which is really about what that dictates sonically rather than eff. per se. 

You like horns, I prefer ESLs which admittedly are not efficient @ 86 dB. But, since I remove 100 Hz down from them they go louder than ---- , which is all I really care about, the ability of a system to reach realistic volume levels.

With horns and large displacement dynamic drivers it's about that as well, but then it's about how realistic volume levels are reproduced rather than merely attaining them. 

At any rate with the amps we have today efficiency is not an issue.

If that was the case it's assuming the amp is the only determining factor in achieving realistic volume levels and overall effortless reproduction, which clearly it isn't. Low eff. speakers will eventually compress both as a dynamic phenomena (as in transiently fairly early on, dulling transient behavior) or more outwardly as a macro-thermally induced ditto heating up the voice coil to such a degree that SPL is reduced from an expected value. 

It only determines volume per watt and not sound quality.  

As an outset, yes, but practically it's not that simple. High vs. low efficiency isn't an all things being equal scenario as there are many differing factors at play comparing the two segments of speakers that will shape the outcome one or the other way. For one, with high eff. and maintaining extension into LF-region comes very large size, and controlling directivity into the lower mids will have the same consequence for the horn size here. The dispersive nature makes a big difference sonically, and high eff. + deep extension is a different meal/animal vs. low eff. and the same. A good quality, high eff. large format comp. driver + large horn combo simply steamrolls over a direct radiating low eff. dynamic driver combo in ways that has to be heard to be understood, whereas a large ESL speaker will have other qualities to bring to the table that in some ways exceed horns, while in others they fall short.