Next level Active DSP Crossover

Hi, here is best/cheap I've found, it's in europe though: https://www.thomann.de/intl/the_t.racks_dsp_408.htm

I think it's only 8 or 9 PEQ per output though, please check manual for details. Sounds fine, no mentionable noise with my horn speakers. Source of noise in my setup is DJ mixer before the DSP, and as said by others adjustable by gainstaging from problematic to non-problematic. There is a fan though, had to swap it to a quiet one and rip out all the metal mesh/obstruction to reduce noise for living room use. A warranty void mod.

Model with FIR costs more. Besides FIR i"d like to have myself more flexibility in terms of routing, would like few programmable knobs to adjust some things and so on, so a Q-sys system would be more suitable I think for flexibility, again with more cost. The t.racks has been rather good basic level DSP, definitely good for the price, it's cheap and fine for what it is, no complaint.
 
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Note that the OP actually disallowed PC-based DSP crossover solutions...and coding via PC, etc. in his original question. I think it's courteous to respond with that constraint in mind, since so many others that I've talked to seem to have the same constraint (i.e., they want a dedicated plug-and-play DSP crossover).
In case this raises a false impression about Q-Sys, or Solaro, or Symetrix...or any of the Open Architecture type processors......
they should defintely be considered as dedicated plug-and-play processors/DSP, imso.

They don't need a PC to run...indeed that's what makes them so desirable, imo.
But they do need a PC make settings, just like about every DSP does (via USB, etc).

The only thing different about them from a tradition hardware DSP, is that the user builds the processing schematic via a PC, before then making settings in the schematics components. They are blank processors before doing so.

Being able to design your own schematic offers unparalleled flexibility.....vastly greater than any fixed architecture typw.
DSP features and capabilities are immense, including FIR.

They don't go obsolete or require a bigger unit, if I/O needs change due to changing a system (like adding a sub or surrounds etc)
Simply make a new schematic with the new speakers. It's easy.
 
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The OP could look for OEM DSP modules, such as these (and I have no idea of their cost and availability presently). Xilica used to make these OEM-type boards about 15 years ago, but stopped 10-12 years ago.

I do know that the miniDSP offerings have been hit-or-miss, and that they apparently do not test their products on very high sensitivity loudspeakers such as modified La Scalas, etc., since they would immediately hear problems with the Flex using its analog (balanced) connections.

I can recommend the miniDSP 2x4 HD, based on use of one presently for my surround loudspeakers (bi-amped Belle bass bins with AMT-1s on top, using unbalanced RCA connections) but you will need two 2x4 HDs in order to handle a stereo pair of three-way loudspeakers, and you will need to route the input cables carefully to avoid common-mode noise. I recommend using the shortest RCA cables possible from you pre-amp or player to keep this source of noise down. Using one of these units for each loudspeaker will provide 3-way output plus an extra channel for a subwoofer for each loudspeaker (i.e., a left subwoofer and a right subwoofer). These units are very small physically, so finding a place for two 2x4 HD boxes isn't difficult.

I'd be very careful of any miniDSP crossover that hasn't been used by someone you know. Their track record hasn't been spectacular over the years--with common complaints of elevated noise floor and very limited gain latitude.

I'm told that the follow-on to the ElectroVoice Dx38--the DC-One--is also very low noise, but they both run at 48 kHz internal sampling rate. These are available on the used marketplace.

You could also look for a good used Xilica XP-4080 or XP-8080 at the right price (...but good luck with that approach since prices on used equipment appear to be rising...).

I think you can see that the price point of Xilica XP series for higher quality DSP crossovers wasn't high for what they are actually doing.

Chris
 
Before you buy something else, have you tried using balanced conductors?

Even though your DSP has single ended outputs, I have had to do this because all my amps require XLR inputs.

I find them quite effective at rejecting noise. I don’t have horns but my midrange is 100+ dB/2.83V.

These easiest way to do this is to buy pre-made long XLR cables, snip then in half, and attach RCA connectors of your choice to each end.

See attached wiring scheme.
View attachment 1265570

Reference:
https://www.diyclassd.com/media/85/b5/1c/1646058855/NC400_04xx.pdf
Page 9
You will want to add 47 - 100R resistors in line with xlr pins 2 and 3 to reduce common mode noise.
 
I've found gain staging to be the culprit behind hiss or hum, far more often than ADC or DAC quality.
Same thing can be said for overall sound quality too.
so, regarding this, over the weekend, I did play with the gain string, turning down the input gain on the DSP, which helps quite a bit. There was still a nominal amount of hiss, but it was barely noticeable.
 
so, regarding this, over the weekend, I did play with the gain string, turning down the input gain on the DSP, which helps quite a bit. There was still a nominal amount of hiss, but it was barely noticeable.
Good for you gigantic, glad to hear that helped.

I often get humbled comparing gear, after discovering that getting the simple audio basics like gain staging correct, meant more to which one i liked best, than the equipment did.
(and humbled again, when I can't tell a difference between equipement when basics are correct, Lol)
 
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You will want to add 47 - 100R resistors in line with xlr pins 2 and 3 to reduce common mode noise.
That won't do anything in that instance. The source is unbalanced to begin with, and adding a small resistor to each leg of the destination will just increase the already high input resistance by a negligible factor.
No performance improvement at all.

Dave.
 
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However, I would like to improve the noise floor, is has a slightly perceptible hum that is present when music isn't playing. It's entirely single-ended, but I would prefer balanced, at least as an option. Finally and this is the biggest issue for me: the DAC is limited to 48Hz, 28-/56 Bit processing, which regardless of whether it's audible or not, effectively nullifies my Schiit Audio Bifrost 2's 192Hz, 24 bit capabilities for hi-res audio file playback.

I'm used to DAWs with 64bit internal mixing path.

But for playback, I’m completely OK with lossless 16 bit 44 Khz.

24/96 and higher is the music industry’s attempt to sell us that same music, repackaged IMO
 
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One of the previous posters included a link to Danville Signal. They have a product called the dspNexus 2/8 which is about $3000. It only supports two input channels (either digital or analog) but has a powerful DSP that can support a lot more than eight PEQ bands if needed. It has balanced inputs and outputs, and uses AKM DACs (currently uses AK4493 DACs, but they are working on a user-retrofittable upgrade to AK4499EX DACs).

I have one of these and have been playing around with it a bit. It uses Audio Weaver software for programming which is fairly sophisticated, but provides a tremendous amount of flexibility.

It only has two input channels though, not the four that you were looking for.
 
I think you're asking for unobtaium with your constraints. The better DSP crossovers define themselves by their low noise and hi-fi outputs, and they simply cost more. Think of this like acquiring tube amplifiers: you might get away with low-cost tube SETs, etc., but their limited output transformer size will come back to bite you.

I'd recommend thinking about this using a different paradigm. If you have extremely high efficiency/high performance horn-loaded loudspeakers, I've found that getting the on-axis SPL response flat to within ±2 dB (using psychoacoustic smoothing) across the board, and also getting the time alignment of the drivers to achieve flat phase to ±90 degrees (again using psychoacoustic smoothing) is a key enabler to a threshold of hi-fi performance that others have only heard using high-dollar electrostatics and certain very high dollar full-range drivers, in addition to all the other positives that fully horn loading brings to the table. Instead of thinking about inexpensive DSP crossovers, think first about getting a DSP crossover that has the required performance, then think about amplifiers and preamps. In other words, stop trying to use cheap DSP crossovers. If you can find one that has hi-fi sound and extremely low noise that also is lower cost, that's great, but I haven't found one yet (and I've tried a few over the years).

In reality, DSP crossover performance for use with fully horn-loaded loudspeakers (including MEHs) is a subject that the low-efficiency direct radiator loudspeaker crowd simply do not understand. The same thing goes for amplifiers having extremely good "first watt" performance--something that I found that many, many amplifiers used by the direct radiator crowd don't have... (Just ask me about my Hypex FusionAmp experience, for example.)


Chris
I realize that I may have an impossible ask with this, but so far, I've had remarkably good synergy building a bang for the buck system that punches well above its weight, in part by diy, (my LaScala/Chorus hybrid speakers, diy mods to amps and preamps) and selecting under-appreciated components that perform beyond their price point and working within financial constraints apropos to my working class means. That's left me with a psychological price point/sweet spot that falls somewhere between $500-$1000.00. To wit, my system currently is composed of varied digital sources, including a MacBook Pro 16, streaming Apple Music, an Oppo BDP-103, Apple TV (which I use rather infrequently) into a Schiit Bifrost 2; A Denon DP-37 TT, with a Denon DL-110 cart and a modded Pro-Ject Debut Carbon TT with an Ortofon 2mBlue cart, into a Schiit Audio Mani 2, which likely will be upgraded to a Schiitt Skoll at some point in the not too distant future. These in turn, are fed into a modded and tube-rolled Schiit Freya + preamp, into a Dayton Audio DSP-408, which is fed into a pair of Alesis RA-150 A/B mono blocks that drive my woofers; a modded Glow Audio Amp 2 EL-84 P/P amp into my mid horns; and currently, a pair of Schiit Rekkr mono blocks, which will soon be replaced with a pair of Nelson Pass Amp Camp mono blocks.
I get that many purists may throw shade at my parsimonious spending habits and the disparate and motley assemblage of components, but incredibly, it works quite well, working within the confines of an average of around $500 per component. so it's within those constraints that I'm hoping to find a solution. Ideally, it would be with something I already own, repurposing the Zoom TAC8 or the Focusrite Scarlet 18i20 that I have in my recording studio, adapting a software solution to my needs. Failing that, using a pro audio device, new or used that offers good performance, less the exorbitant price tags of the likes of xilica, ashly and others mentioned above that still go for prices well north of $1200 on the secondary marketplace.
 
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...I get that many purists may throw shade at my parsimonious spending habits and the disparate and motley assemblage of components, but incredibly, it works quite well, working within the confines of an average of around $500 per component. so it's within those constraints that I'm hoping to find a solution...
PWK actually made a point of using lower cost drivers and other components to minimize costs...in order for his loudspeakers to remain affordable. There is a reason why the types of speakers that PWK built (and the various variations on a theme that go along with that) can sound so good--and it had to do with knowledge and ability of the guy that did it. I have always identified with this. [The people that now run his company no longer have that goal: they seem to relish raising prices to as high as they can--like other audiophile boutique firms--and punish their customers by ignoring their needs after buying in, like no other company I've ever seen in my part of the world. PWK's culture is apparently gone.]

You can usually tell engineer's setups from audiophiles: most audiophiles want to "buy perfection", while engineers always seem to economize greatly--but the sound quality is second to none. And the price of bought electronic gear is usually the place where most audiophiles waste most of their money. Put the money where is counts, and only where it counts.

But I've learned there is a limit to how cheap you can go before you start to throwing good money after bad. I've found with DSP crossovers that too many try to buy poorly designed boxes that only were meant to be used in after-market auto sound systems--where hi-fi really isn't the most important requirement.

Chris
 
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I'm not sure what your application and goals are, but the manual suggests that the number of taps might be fairly limited if you are thinking of FIR-filtering low frequency signals.

True that. The FIR capability looks mainly like marketing, especially given that the unit runs at 96kHz.
Given taps are a limited hardware-processor resource, a lower sample rate can do more work than a higher one.

But the Thomann processor does look very interesting in terms of good affordable functionally.

One answer I would need to know though.....is what is the smallest time delay increment available?
Anybody got one?
 
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The Symetrix DSP I have supports maximum of 1024 taps for FIR - and can do this for 8 channels. Then it is out of capacity. I plan to use it for 4 way stereo crossover and EQ as needed - and there is more than enough capacity for that. You can easily evaluate the capabilities in the designer software which is a free download.

I have only used the smallest DSP by t.racks (I have three of them) and it is very easy to use - each model also has its control software which will let you evaluate the possibilities.
 
@mark100 It samples at 96khz, so I would imagine the smallest delay step would be about 10uS.

Dave.
Thx Dave,
Yep that's what we would hope. The problem I've found with delays is 'coarse' vs 'fine'.

When i see specs giving a maximum delay only, and in coarse units, like the Thomann unit's manual says 0-680ms,
i've encountered that there can be an inability to fine tune delay down to the sample.
It's as if delay input fields only allow a limited number of significant digits.
It's bit me in the butt before, so it's something i question in lower priced units.
I've had to string two delays together, one for 'coarse full ms', then one for fine....if such serial delays are allowed, and if sample delay resolution is available.