Is it possible to cover the whole spectrum, high SPL, low distortion with a 2-way?

Sorry fluid,

I just went wow, that's very low volume casual listening for me.
No apology necessary and we are all free to listen at whatever level we choose.

Here is a quote from Bob Katz, (link below) that I think could explain why that is a level that I feel comfortable at and can optimize my system to.

Adjust the monitor gain to yield 83 dB SPL using a meter with C-weighted, slow response. Call this gain 0 dB, the reference, and you will find the pop-music “standard” monitor gain at 6 dB below this reference.
By now, we’ve mastered hundreds of pop CDs working at monitor gain 6 dB below the reference, with very satisfied clients.


Level Practices (Part 2) - Digido.com

And the idea that the TC9 line can produce 10Hz just isn't at all realistic, it barely pulls much lower than 100Hz, imo/ime.
It is not an idea it happens, I can see it, feel it and hear it when the house structure starts to vibrate with it. In no way is it flat at 10Hz but the frequency is produced with enough SPL to cause trouble if not managed.

I have no simple way to prove it to you if my word alone is not enough.

That's could be as much as 12 dB less than what I would say is valid and that is dB(C) fast.
I'm not sure I understand what this means could you expand on it a little.

My mic is calibrated for frequency but not SPL so I have to rely on a cheap meter to get something to compare to which is why I have the conditions.
 
Here is a quote from Bob Katz, (link below) that I think could explain why that is a level that I feel comfortable at and can optimize my system to.

Adjust the monitor gain to yield 83 dB SPL using a meter with C-weighted, slow response. Call this gain 0 dB, the reference, and you will find the pop-music “standard” monitor gain at 6 dB below this reference.
By now, we’ve mastered hundreds of pop CDs working at monitor gain 6 dB below the reference, with very satisfied clients.


Level Practices (Part 2) - Digido.com

thanks for the link, I laughed because around 83 dB is what I have measured in my listening room with Behringer DEQ 2496 + ECM 8000 in 2005

(the thread is archived and cut but the number 83 dB survived: max SPL a wymagania sprzętowe - DIY - Audiostereo.pl )

Running a sound pressure level meter during the mastering session confirms that the ear likes 0 AVG to end up circa 83 dB (~86 dB with both loudspeakers operating) on forte passages, even in this compressed structure.

I can confirm, my ear likes it too.
 
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^Ty so much for the link...40db headroom aye..,


For the hardware i think you'll gain more from Mark and Fluid:

Anyway i haven't changed my mind since last time we talked about it: pc/soundcard is the most powerful and economically rational choice in my view.

Thanks for the advice

Pc + sound card isn’t the full pic Acourate doesn’t host filters and neither do sound cards
 
Pc + sound card isn’t the full pic Acourate doesn’t host filters and neither do sound cards
For me the missing link is to use Jriver as it can process the multiple convolution files needed, as well as VST plugins etc.

You can use any other VST host or DAW if you want to although it has been a long time since I have gone down that road, for music playback Jriver does everything I want and I am not that easy to please :)

If you had DSP hardware you could also run crossovers, basic driver EQ and delays in that and then run stereo convolution over the top.
 
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Fluid is right Jriver is an answer.
The issue is to integrate it into a 'professional' environnement: either you have a dedicated pc running it either you start messing with 'virtual cables' and most incertitude OS related ( don't ever let Windows take control of audio flux!).

As you'll be running a DAW there is another option (which i use so can testify it works even on a prehistoric pc running win XP and hardware 20 years old):
you setup a deconvolution plugin on your master/monitor section within your DAW.
What is a deconvolution plug in you'll ask?

Any convolution reverb which accept external Impulse Response: this range from Waves IR1 and the likes to Voxengo Pristine Space ( this is the one i use in the pc i'm talking about: it is multichannel and low on option to messup the IR and run on XP).

Of course all this references are 32bit and not at all uptodate but you'll easily find equivalent.
Maybe take a look at what amp modelers plugins ( guitar amp modeler) can do because it seems to be the new rage within guitarist ( "i've got IR from 'Steve Vai'* cabs! Let's have THE sound", "well no mate: you bring an amp and cab and we mic it. Otherwise you work with someone else...").

* place any guitar hero name there.
 
If I use my usp4 cards to do the linear XO.....yes I could take Acourates filters and load them into IR-1....thats what I pictured as one solution for FIR, for me, all along....I just never put two and two, together....that is.....I can use Acourate just like I've used Rephase in the past BUT....in the terms of using Acourate to generate linear XO filters.... I have no where to go with that...Jriver is for playing back media? You run your XO crossovers through Jriver? Jriver is going to be a system wide crossover for me? I didn't think that thats what it did??
 
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thanks for the link, I laughed because around 83 dB is what I have measured in my listening room with Behringer DEQ 2496 + ECM 8000 in 2005
....
I can confirm, my ear likes it too.

Hi, in my experience it is relative: room and style come into play.

The 83dbspl reference is the standard within movies/ theater setup ( Dolby and other norms use it as base).
When in a theater i usually find it ok but the rooms are usually big and everything is mastered: the loudspeakers are setup and won't move, everything is calibrated, acoustic treatements are localised for the kind of multichannel used, these are room with high ceiling height and seat have a coeficient of absorption which is the same as a typical body so acoustic doesn't change depending of the crowd.
All this lead to find the spl confortable.

In smaller room i usually find 83dbspl to be too loud for the kind of music i listen to ( they most of the time belong to K14 or K12, in other words typical 'pop' dynamic range).

For classic or jazz i think i could have that 83dbspl ref and like it however but for other reason: this is related to background noise. Typical domestic space have something like 50dbspl background noise so when you introduce 'high' dynamic materials you'll need to up the level a bit.

What i miss the most from professional control room is not the acoustic treatments but soundproofing: very quiet place.

Whith lower DR material i suspect our brain have a defense reaction in smaller room: to high an average level and our brain needs to protect the interface to the world so it tells us to lower volume.
I suspect this is ER related as i don't have this with headphones. But all this are speculations from my side.
 
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I just never put two and two, together....that is.....

You fear about running multiple instance of same reverb's plug ins?
Come on Camplo, 2001 is long time ago: all DAW no incorporate plug in delay compensation. You won't run into issues about this particular point ( the latency induced by software may be another question but i'm sure it should not be a deal breaker).

I use Acourate just like I've used Rephase in the past BUT....in the terms of using Acourate to generate linear XO filters.... I have no where to go with that...Jriver is for playing back media? You run your XO crossovers through Jriver? Jriver is going to be a system wide crossover for me? I didn't think that thats what it did??

Jriver have a 'live input mode' ( or whatever it's name) to allow streaming of broadcasted material ( from tv, radio,...).
Within the software you have deconvolution engine which takes care of the filter ( IR whatever it is FIR or IIR).
 
OK I think I finally got it lol!....you said it once already it just didn't sink in...

XO...I don't have to have linear XO if I'm going to fix it(phase) elsewhere...I could use a minidspDirac/DEQX to do that or.....or any Convolution hosting hardware/software to do this.....
 
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Me neither, never used Accourate so can't speak of it. I think they had major update in the previous years and it may take everything in charge now ( integrate deconvolution engine?).

From what i've read member Lewinski01 use it, try to ask him.

Video game with FIR won't work as you'll have latency from treatment (+ eventual other source within the digital stream in computer/soft): IIR is the answer.

Do you need pristine accuracy of sound when playing 'Pong' or 'Tetris' and it's 8bits soundtracks?
Wait, do have videogames made progress to the point it is not big pixels and 8bits soundtracks anymore?! Omg i should quit my cave from time to times...and bought something else than an Atari or a first gen Gameboy! :D

Arcgotic: is EqAPO Asio compatible? If not forget it, in pro situation you can't rely on Windows for the digital audio stream.
 
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You can use EqualizerAPO under Windows. All sounds from all aps go through EqAPO. Create impule response filters for crossover in Rephase. And load them in EqualizerAPO (FIR, IIR whatever you create in rephase)
I had latency problems with EqualizerAPO and looong time ago never touched it again...maybe I should give it another try.

Video game with FIR won't work as you'll have latency from treatment (+ eventual other source within the digital stream in computer/soft): IIR is the answer.

So am I right that the minimum phase errors can be fixed downstream with convolution...it doesn't have to start at the XO...I should say....so theres no benefit of starting at the XO?
 
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No: FIR introduce latency, this is the nature of the beast.
To have frequency and time domain events treated separately you need latency.

You can't go back in time ( i would love this option just to remember how it is to have hair and no needs for glasses to read as i have my first pair of 'old man' glasses in two weeks! :D ).
 
you did not answer my question at allll lol
So am I right that the minimum phase errors can be fixed downstream with convolution...it doesn't have to start at the XO...I should say....so theres no benefit of starting at the XO(with linear phase)?

maybe I should reword that....

linear phase xo and linear phase correction downstream.....
vs
minimum phase XO and linear phase correction downstream...

samething?
 
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I did but it wasn't plain words ( and now i'm old you'll have to deal with me with respect and issues associated: babbling, outdated references,... :D ):

Why do you want to treat minimal phase issues with convolution? Frequency related issues can be dealt with IIR eq ( no need for convolution here) and i'm sure you can live with IIR filters for gaming and tracking.
So for minimal treatment latency you could use a chain of plugins ( eq and filters on your Daw Control Room section).

If you choose to use convolution well you integrate everything in one treatment but i don't know if it'll be as fast as dedicated plugs ( i never compared it). From an ergonomic point of view it'll be easier though ( you load dedicated preset rather than loading a whole project).

It is difficult to have a definitive answer as there is a lot of variable to consider and in the end will depend on your workflow.
 
is this my fault or yours lol!

linear phase xo and linear phase correction downstream.....
vs
minimum phase XO and linear phase correction downstream...

same thing?

Crossover; If I use minimum phase high order slopes (48db) the group delay is high....I can fix that downstream with Acourate/rephase filter....you put the filter into the convolution software/hardware ie Waves IR, Jriver, miniDspOpenDRC-DI, random hardware IR reverb, etc etc)

I can also use outright a linear phase crossover......
(Im not talking about voicing...I'm speaking strictly on the hp/lp filters)

If you choose to use convolution well you integrate everything in one treatment but i don't know if it'll be as fast as dedicated plugs ( i never compared it).

I guess this is the answer...and the Question...anyone know?
 
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I guess in my case, still...when wanting best latency I naturally have to pick IIR...so thats settled but When I'm wanting best SQ...will I land at the same destination regardless of latency, looking at those two paths???

linear phase XO(hp/lp) and linear phase correction downstream.....
vs
minimum phase XO(hp/lp) and linear phase correction downstream...