DSP Active Crossover

if your your active drivers need impedance flattening, notch filters or if the speaker has baffle step compensation you are fighting an uphill battle trying to use only DSP to address those issues.

@Avanti1960, having built both, I can say this is not at all the case. In all cases, impedance flattening circuits such as a Zobel, increase power required for the sake of making the crossover work closer to an ideal state, like it would with a resistive load. I can categorically state this is completely unnecessary with a DSP based amplifier and crossover. It’s one of the major benefits of designing active speakers that you can ignore the impedance of the drivers.

Another way to say this is that in a passive speaker I only care about impedance flattening because of the effect rises and peaks can have on the frequency response. With DSP, any such issues I can deal with directly in the EQ.  THe point is moot though because with an excellent plate amp the impedance curve of the drivers just doesn't matter (so long as it's high enough).

There are hybrid speakers, which use both active and passive crossovers but I think these are becoming rarer with the common availability of 3-way plate amplifiers. IMHO, and not all speaker designers will feel this way, there’s no upside to a hybrid system if I can go fully active.

The baffle step compensation you reference is a frequency domain issue as well which is quite easily dealt with by a DSP EQ instead of with additional passive components.

Another great advantage for the designer of a fully active instead of a hybrid system is the ability to digitally delay each driver independently and achieve a quasi point source output with high order filters giving you most excellent on and off axis response which you’d have trouble with a hybrid.

The flexibility of a DSP based crossover sometimes causes bad choices though, such as picking bad sounding or poorly matching drivers and then hammering them into shape with EQ, as well as using global EQ to fix bad crossover choices.

You sound ready to take the next step in building speakers, so I really hope you get excited and build some for yoruself soon, either passive or active.

I would be willing to bet that most active speakers with DSP still have some degree of passive crossover networks prior to the electrical connections on the speaker.  many drivers require a simple impedance flattening or notch netwok to rid the driver of response peaks. 

if your your active drivers need impedance flattening, notch filters or if the speaker has baffle step compensation you are fighting an uphill battle trying to use only DSP to address those issues. 

The minidsp SHD (2 way digital xover) uses an ESS 9028 DAC chip......pretty good. The Audiophile Junkie (see his Youtube channel) uses this Minidsp SHD in his super system. I posted this on a couple of other threads.....but I cannot stop talking about this....so here goes again.

I have been quietly following this thread. have a feeling that the OP will be thoroughly disappointed with the minidsp SHD since he appears to be used to a Denafrips dac and whatever preamp he’s got.

For my attempted application, i tried to replace the Denafrips Hermes+Venus and a Yamaha C5000 with the minidsp SHD in pursuit of an active crossover. The SHD is awful in comparison and i just got rid of it.

To each his own....(with the theory crafting n all).

@simonmoon wrote:

I’ve heard several systems with the mini DSP, and have been unimpressed.

I’ve also heard several very high end DSP systems, including the Kyron. And that system kills! Even Mr. analog, Michael Fremer loves it.

But my overall evaluation is, that it takes a very special DSP speaker system to best the best passive system.

It’s not as much the DSP unit as it is the implementation. My IIR-based DSP unit cost me just over just $1,000, and it’s an excellent piece of equipment for what it’s supposed to (make me) do, and one that also holds up perfectly well to Lake units (costing much more). That’s for nothing however if you don’t know how to turn those settings into proper, audible effect, of which there are different routes for that to be accomplished. I still prefer setting DSP-values manually by ear with the aid of measurements (and input from friends), and while a painstaking and lengthy process the results can be, and actually are extremely good.

Of a range of passively configured speaker setups I’ve heard that were converted to outboard active configuration - that is, bypassing their build-in passive crossover completely and replacing them with line-level DSP’s/electronic XO’s and more amps - each and every one of them eventually saw a substantial upgrade in sound quality over their passive iteration (and that was obvious fairly early on), to everyone listener involved. That’s all I need to know and a testament to the potential of active configuration, not least from the important basis of comparing the same speakers with different filter configurations.

Many are, on principle, against DSP due to speculated, negative effects of A/D-D/A conversion steps with analogue inputs only, but with no experience to really speak of that would actually single out this particular aspect as the detrimental factor. Why even make any assumptions as to what may or may not, technically, be the reasons for a speculated deficit?

One of the best setups I’ve heard, an outboard active one at, comprises the exact same DSP unit I’m using. For anybody wanting to tell me it’s an IIR-based filter and not a FIR ditto, while implying perhaps it’s the lesser solution, I can only stress the importance of seeing the forest for the trees in actually listening to a properly implemented active setup and let your ears decide. Should the FIR-filter hold the upper hand sonically, which theoretically it does not least in being able to generate linear phase response, that’s only an added bonus that will potentially distance active from passive even further.

Most systems that use the Minidsp stuff are low end systems. I don’t know any high end speaker company that uses them (anyone?)......however, the latest Minidsps are very good. If you are just making a 2 way speaker like I have described then the ultimate version (using regular analog amps and not digital amps) would be using the digital only minidsp that costs $500 and add a better power supply and run out of it digital into the DACs of your choice. Use a serious DAC on the mid-tweets and a less expensive one on the woofs. This would compete against lots of super high end speakers. The DACs in the latest Minidsp are good but not great. Same with the dspNexus......just AKM DAC chips and op amps.....the same as $500 separate DACs. The dspNexus has the advantage of having more channels if you want to tri-amp or bi-amp plus subs. It would be great if there were a version of the dspNexus that had digital outs....so any system using it could be improved with better sounding external DACs. The Minidsp SHD has both analog outs and digital outs...but, it is a two way only.

See my post on 9-22 above for the link to my webpage that describes all this (recently updated).

I've heard several systems with the mini DSP, and have been unimpressed. 

I've also heard several very high end DSP systems, including the Kyron. And that system kills! Even Mr. analog, Michael Fremer loves it.

But my overall evaluation is, that it takes a very special DSP speaker system to best the best passive system. 

Hello,

I am hal2010, AKA Richard Hollis.

If you have any questions about DSP crossovers for the dspNexus 2x8 system, will be happy to answer them.

Best regards,

hal2010

@dsp - Al, I didn't mean to imply that the dspNexus isn't a well engineered product.  I would expect that replacing the SMPS with a linear supply that was good enough to provide a noticeable improvement to the sound quality would more than double the cost and significantly increase the size of the enclosure (also increasing cost).

As is, the dspNexus provides a lot of value for it's price point and is very nicely constructed, allowing easy upgradeability to the DSP and DAC modules. 

I look forward to meeting you at CAF. 

Hi Everyone,

I would like to discuss the choice of the switching power supply in the dspNexus. Commercial switching power supplies solve a lot of practical problems when converting AC mains to low voltage supplies in a commercial product. They are not ideal.

I evaluated 4 different switchers during the development process of the dspNexus. Most suffer by having insufficient current capability with the negative supply. This was clearly audible with the headphone amplifier even with 100 watt units. The one I chose was good in this regard. It is the only board in the dspNexus that I did not design.

That said, a large linear mains supply is not a panacea. There are three linear supplies between the DACs, ADC and Clocks. These are not all LDOs since high frequency power supply rejection is also important.  The goal is to have a low noise, low impedance supply located near each target device.

Designing good audio products is a process of making many small decisions. Collectively, the choices you make determine the outcome. I started making high end consumer audio products 45 years ago as an analog engineer. I have used DSPs for 30 years, so I have a bit of experience.  I welcome discussing design philosophies and I hope many of you will consider the dspNexus as a good solution for your systems. I will be at CAF and I hope to meet many of you there.

Al Clark
Danville Signal

 

@ricevs - I have been using the dspNexus as the DAC/preamp, so I'm feeding it USB audio directly (no conversion from analog). I don't think it makes sense to use my Terminator + and preamp ahead of the dspNexus. 

I agree that the dspNexus could be improved. I think the biggest weakness is the power supply - the simple SMPS that is currently used could be improved a lot with a good linear supply with shunt regulation. I think if I decide to go this route, I will probably implement my own power supply. 

The biggest advantage I see to use a DSP crossover is to be able to implement steep filters with no added noise. If I implement analog active crossovers, the steepest I'd probably be able to get away with is 24db/octave. This is still better than the 12db/octave passive crossovers I have now though. 

I agree there is a bit of room for improvement in my current crossovers, but this is clearly diminishing returns. Still, I may consider it at some point. I've just got too many projects going on in parallel right now, and I'm having trouble actually finishing any of them 😀.

As far as facing all the drivers in the woofer towers forward, I have not been able to discern a difference. I had them facing forward in my triple towers and I think the quad towers are just as tight and clean sounding - probably a bit more so - than the triples. There isn't an easy way to turn them around anyway. All the holes for the wires are only drilled on one side. 

With such a tall tower, I'd be a bit nervous about having all the drivers facing forward since the weight distribution would be heavily toward the rear. Even the triples were slightly tippy. 

BTW......I am pretty sure you would get much cleaner bass if all the servo woofs were all facing forward.  Does not make sense to have two facing backwards.  Even Rich Hollis is doing it the way I suggest....so did GT Audioworks (who now uses other non servo woofs....but all facing forward)

https://positive-feedback.com/show-reports/lone-star-audio-fest-2023/

Jaytor,

Given how good your DAC, preamp and passive crossover are......I can see moving to a stock electronic xover as being just a side ways or lowering quality move.  The analog input and output stages are just op amps in the Danville (nothing like the analog stages in your preamp!!!).....and how good are the power supplies and routing and wires and jacks?  It is not made anything like the preamp you made or the DAC you have.  You are removing an excellent passive simple crossover for a way more complex and not as well done thang.  A better A/B would be to use the Danville as the DAC and preamp, as well.  Therefore you are not as much adding something but just changing.  You would then have no external DAC, no external preamp, and no analog cables between them.  You just run the digital signal directly into the Danville and then through their DAC and output stages and then right to your amps.  This is how Rich Hollis is using the Danville.  He uses no preamp or external DAC....he gets some pretty darn good sound.  And if you modded the output stages of the Danville you would then get even better sound.

This is why I am such a fan of digital amps.  You have no DAC, no preamp, no analog cables and you also have no analog stages on the output of the crossover or normal analog amps with feedback (or no feedback).  The digital signal from your source goes directly to the chip in the digital amp and then the PCM is converted directly to PWM and then drives the GaN output stages that drive your speakers.  By using two stereo digital amps you can bi-amp using the Minidsp Flex of SHD and you can build a two way speaker that is out of this world transparent.....because it has no analog crossover and has these low distortion digital amps (sonically low distortion)......all drivers mounted on open baffle and driven directly without connectors from the amps.....this is unheard of simplicity.  Of course, you have a 3 way speaker system with servo amps so going this way would make it way more complicated.....so this would not work for you.  I am sure you have seen my webpage that describes what I am talking about.  This is world class amp/speaker system for $6K.  If you wanted to go bigger you could run 6 12 inch woofers per side on their own baffle and run 8 planars in a long line source on another baffle fo another $2K.  Imagine the impact and transparency of such a monster.......and still only $8K or so.  We are talking 6 12 inch woofers and 8...eight inch planars per side.....open baffle, fully equalized and time aligned.....would blow you away...........just like your current system.....but look at the cost difference!  The only thing you would not get with such a system is super low bass like with the servos.  

Here are a few suggestions to make you current analog xover better.

1. Change the foil coil on the input of the midrange to a 12 gauge version (looks like 14 gauge? is what you are using?  Also, you need to go into the inside of the coil (all coils you use) and out the outside....or the outside to ground if used as a shunt.  I like the Jantzen 12 gauge wax paper coils.  Mundorf has some newer foil coils that are damped as well.

2. Change the Erse coil to a copper foil coil.  Even in a compensation circuit it still makes the same difference.  Erse coils are veiled....big time.

3. Change your speaker input connectors to WBT copper nextgens and cut off most of the solder tab in the back.  Then remove those massive speaker banana posts from your speaker wire and form the end of your speaker wire into a loop and tin it and wrap it around the WBT jack or make your own tinned hook out of 14 gauge VH audio cryoed ofc wire that you use as a spade.  This is much more transparent than what you have.  Even better would be to hardwire your speaker cable directly to the wires in the crossover....or use my plastic clamp system as shown on my Peachtree GaN 1 page.  I have not used any speaker connectors on any speaker or amp in over 40 years.

You could also try my Ground Enhancers and Music Purifiers on the input of your xover....mucho better sound for practically nada moolah.

Whatever you do....please....have fun.

@ricevs - Unfortunately I haven't had a lot of time to work with this yet. I've been buried with work (I run an ecomm business and do all the coding - getting ready for the holidays) so this has taken a back seat.

Setting up the dspNexus means I have to disconnect my passive crossovers,  set up an extra pair of amps and get everything plugged in, and change the settings on my subs, so takes a while. And then takes a while to switch everything back. So I have to be able to dedicate at least 4-5 hours to make it worthwhile and haven't had that much free time in one chunk in a few weeks. 

Once I get into November, I should have more time since I don't like to make changes to the websites during the holidays. I'm planning to attend CAF next month and look forward to discussing my setup with Al Clark and Richard Hollis. 

So far, I have gotten a simple crossover setup to work, with a limited amount of equalization to the subwoofer channel. It sounded ok, but will clearly require more filter tweaking before I can decide whether I like it better than the passive crossover.

Jaytor,

How is your active system doing?  How does it compare with the passive version?  Any tradeoffs?  My friend in Panama owns the very first Line Force speaker that Danny made and was made from Aluminum.  I just sent him a modded Peachtree Gan1 amp to use on the panels.  He modded the passive xovers with my suggestions.

I don't understand how you were using a Tact for bi-amping.  Does it have analog line level outs?  Or is this a digital out for subs?  Or two sets of digital outs?  And how were you using the SHD?  Analog outs or digital outs?  And if you are using digital outs what external DACs were you using?  Graininess in what configuration and compared to what?

@ricevs ,

No, you can not parallel SHDs. The Dirac Live only knows about 4 channels in the SHD. As far as the crossover are concerned you might be able to double it up although I think that is a silly proposition. You are much better off with Dirac Live than a three way crossover. It is hard to explain to people what total control over a system/room's amplitude response means or sounds like. You have to hear it. 

@ricevs 

When my TacT 2.2x bit the dust the DEQX was still two months away. So, to fill in the space I got a MiniDSP SHD $1500 with UMiK 2. The Pre 8 is still one month away, so I will be glad to tell you what I think after a week with the DEQX. All I know about the processor at this point is that because of Covid and delays in getting the first processor they redesigned it for a new and better processor that became available. It is a 64bit floating point system. It also shot the price up a couple of grand. My 25 year old TacT sounded better than the MiniDSP. Lets face it the MiniDSP has 4 DAC channels and two ADC channels, It also streams and handles DIRAC Live. That is a lot of stuff for $1500. Benchmark Media Systems is using an SHD Studio and two of their own DACs and think it is great. That is a $6000 proposition. That is a tempting second choice if the DEQX is too rich. The SHD has a slight graininess to it. It is not crystaline. The TacT was, and as long as you had a computer hooked to it you could do anything and I do mean anything. I am not sure yet how a computer relates to the Pre 8. I do not even have a manual yet and yes, I asked for one.

mijostyn, Did you A/B the minidsp SHD with the latest DEQX in the same system using all the same gear at the same time? "clearly the best" can only be known by serious listening tests. If you did do a serious A/B.....what were the sonic differences? The Minidsp is $1300 with standard mic.

The SHD also has digital outputs that can drive the fantastic and super transparent modded Peachtree GaN 1 amps. A bi-amp system using two modded Gan 1 anps and the SHD should be super duper and cost less than $5K.....so all you need is drivers and a baffle......the SHD can take analog inputs (like a phono stage), is a Roon ready streamer has digital inputs including usb drives and has analog outs as well as the digital outs....all volume controlled. Super versatile.

Of course, the SHD is only a two way xover.....so complicated systems need something else unless one can daisy chain or parallel the minidsp units. Don’t know if that can be done.

@mijostyn - that looks like a nice piece of gear. Do you know what DSP it is using?

I think the Danville dspNexus 2/8 is an excellent value at $3K, but currently doesn't have software as simple to use as products like the DEQX. You have to be fairly comfortable with computers.

The HW is well built, but the casework is not as fancy as the DEQX Pre 8. The power supply seems to be the weak point and I suspect the DEQX is better (based on what I've seen for their previous products).

I was able to try out the dspNexus on one channel of my main system yesterday. So far, I have only set up the crossovers (96db/octave), adjusted time alignment between my main planar line arrays and my woofer towers, and added a bit of PEQ to the low-frequency channel. My bass response is fairly smooth without any equalization so I didn't apply a lot.

I haven't tried to create any convolution filters yet. And so far, I've only played it from my PC, not from my streamer. Next step is to set it up connected to my streamer and playing both channels to see how it sounds. 

@jaytor ,

Good move. I have been doing this for 25 years and keep a close eye on the market. I have been chosen to beta test the new DEQX Pre 8 which has a 4 way digital crossover you can program to do whatever you want. It has complete bass management, room control, EQ capability at 1 Hz intervals (target curves) and streaming capability. It will be $10,995.00 retail. There will be a less expensive unit following called the Pre 3 which has a two way crossover most people will use for subwoofers. It should be about $4000 less expensive. Another approach which would be much less would be the MiniDSP SHD studio and two good DACs like the Benchmark. That will run about $5000. The MiniDSP SHD (not the studio version) has its own internal DACs and costs $1500. Its crossover is suitable for subwoofers. It uses DIRAC Live which is easy to use and works well, but it is not very flexible and the EQ only has 10 corners. For the money it is excellent. There is the Anthem STR preamp for about $4000 which I have not had the opportunity to use. Finally, the Trinnov Amethyst which has great room control but only a very basic subwoofer crossover. The DEQX is clearly the best unit for 2 channel stereo on the market today and if all you need is a single 2 way crossover the Pre 4 is going to be a bargain. I expect it will be released in about 6 months. 

I use DEQX to actively crossover my subwoofers at 85 hz. It also corrects timing and room issues.  It is the only way I can integrate subwoofers. Fabulous results but I needed professional help from Larry Owens — say approximately three hours of our combined time. 

I've read a lot of posts over the past few years from audiophiles that have switched to active crossovers and I can't remember any cases where the poster didn't feel switching to active was an improvement. It's obviously more complicated, but has some obvious advantages. I'm looking forward to trying it out. 

I can buy everything but the skill set.  I will buy manufactured products and keep all of my fingers. 

The minidsp SHD (2 way digital xover) uses an ESS 9028 DAC chip......pretty good. The Audiophile Junkie (see his Youtube channel) uses this Minidsp SHD in his super system. I posted this on a couple of other threads.....but I cannot stop talking about this....so here goes again. This system will blow your mind to pieces and for practically nothing:

http://tweakaudio.com/EVS-2/DIY_Bi-amped_super_speaker.html

A two way system as described is actually very easy to  set up and configure.....As I say in the article.....even a 10 year old can do it.

@jaytor wrote:

Danville provided a sample 3-way crossover design which I have modified to meet my needs, although I’m sure I’ll make a lot more changes before finalizing my design. I’m starting out with the crossover points set at 180Hz and 1800Hz, using 8th order (48db/octave) filters. These are the approximate crossover points that my passive crossovers provide, although with much more shallow slopes.

The Audio Weaver software is very flexible and powerful, but requires that the user set all the filter parameters. In other words, you can’t just hook up a measurement mic and have the software automatically determine all the filter parameters to match a target curve. There is a lot more trial and error, but at the same time, you know exactly what processing is being done.

I have to make some speaker patch cables to connect between my speaker drivers and amps (bypassing the passive crossover), and then I can start testing on my main speakers. I’m hoping to do this over the weekend.

It will be interesting to learn of your findings when finally bypassing your main speakers’ passive crossovers for a fully active configuration. Not many do this on this forum, i.e.: switching their passively config. speakers to active ditto, nor those who start from scratch with speakers that came sans passive crossover to begin with. You don’t just buy speakers at hifi retailers without passive crossovers as an intended package for outboard active config. - except the rare likes of JBL M2’s, Sanders Sound and a very few others - indeed for that you’d typically have to convert your passive speakers into active, go the DIY route or pro segment. The latter two is what I did with DIY (shared) sub designs and pro cinema speakers intended for active config.

My main inspirational source for going active is a friend of mine who converted his passively config. S.P. Technology Revelation speakers to active ditto, and as it eventually turned out that switch was quite the, well, revelation. The passive S.P. Tech’s were power hungry beasts as few and developed on the Crown Studio Reference I amps, which are very powerful with a crazy high damping factor in the lower regions. However very few hifi amps can muster up that kind of speaker control and power delivery, and you’d wonder why the S.P. Tech’s didn’t come with the mandatory recommendation of being paired up with the likes of Ref I amps they were developed on. The bottleneck though proved not only to be about power requirement, but that bypassing the passively crossovers entirely - upon thorough implementation to active after a lengthy process - simply meant a massive upgrade in overall sound quality. Initially it was still very obvious that going active was the right thing to do - that was apparent from the very get-go.

Myself I’ve now used about two and a half years steadily upgrading my Xilica DSP-based, actively configured speaker setup in stages, lately only a few weeks ago. Detailed factory specs from EV manuals were initially involved to get a bearing on the crossover point (2-way main speakers, + subs) and where to place some of the filter notches; soon after near field measurements came to the aid for more precise adjustments, and what followed was a bit of restructuring acoustic room treatment, main speaker placement, some horn damping, amps experimentation (using similar amps top to bottom is a must), and eventually the lengthy process of making filter adjustments by ear mostly involving delay and subtle PEQ corrections. Gain structure, filter slopes and types (36dB/octave L-R here) and crossover points are usually found and settled on fairly early in the process, but lately we found some rather critical improvement in overall coherency with asymmetrical crossover points over the main speakers and delay reconfiguration incl. the subs as well.

Some may find it an impediment with the active speaker approach and setting filter values by yourself that it’s likely a never ending process, but once you settle on the rougher structure of your speakers, amps, source, cables and overall rudimentary implementation, what follows is the process of fine tuning all that into a progressively coherent whole in your acoustic environment in ways of specificity and accuracy that a passive speaker setup simply can’t equal. It’s not that much about swapping hardware any longer and being on that merry-go-round to find the perfect matches component-wise, but rather seeing the gains that can be made from a much better outset with a DSP acting as a digital crossover, and the many options that are offered here implementing the existing hardware components. To me at least that makes much better sense.

I've made some progress with the dspNexus. This product is still considered an "early adopter" release. The main limitation is documentation. But I've had a couple of video conferences with Emilson, Danville's software engineer, and now have a pretty good feel for how most stuff works. 

I bought a cheap active speaker to use for testing so that I didn't risk damaging my main speakers while I learned how it all works. This has allowed me to verify that the filters are doing what I want. 

Danville provided a sample 3-way crossover design which I have modified to meet my needs, although I'm sure I'll make a lot more changes before finalizing my design. I'm starting out with the crossover points set at 180Hz and 1800Hz, using 8th order (48db/octave) filters. These are the approximate crossover points that my passive crossovers provide, although with much more shallow slopes. 

The Audio Weaver software is very flexible and powerful, but requires that the user set all the filter parameters. In other words, you can't just hook up a measurement mic and have the software automatically determine all the filter parameters to match a target curve. There is a lot more trial and error, but at the same time, you know exactly what processing is being done. 

I have to make some speaker patch cables to connect between my speaker drivers and amps (bypassing the passive crossover), and then I can start testing on my main speakers. I'm hoping to do this over the weekend. 

So far, I have no regrets going for the dspNexus. It seems like a well-engineered and powerful device. 

I have had good luck with a DBX Drive Rack PA 2....am considering upgrading to the DBX VENUS 360....both reasonably inexpensive . Excellent software and Smart App to control all functions from your Smartphone.

The system is a Tri Amped Horn system.....

Please keep us updated . I’m especially interested in the learning curve, flexibility, and functionality of Audio Weaver.

For my  filers, I spoke to Bruce Thigpen at Eminent and pretty much copied what he suggested. For his next gen speakers, he is using DSP, so is aware of what works best. Yes Audio Weaver is what was used to implement the filters. I’m not aware of any other option that is compatible with the DSP Nexus.

@lewinskih01 

I received the dspNexus a few days ago, but I've been buried with work so haven't had a chance to do anything with it yet. I've got a measurement mic on order that should be here early next week. I'm hoping to find some time in the next several days to at least hook it up to a computer and download the Audio Weaver software.

Thanks for the info. I found Mitch Barnett's book on Amazon. I'll check it out. I'll also take a look at the Acourate software to see if this will work with my dspNexus. 

@jwr159 

What software did you use to create the filters for your dspNexus? Anything other than Audio Weaver?

Finally got my act together.
 

Since purchasing the DSP Nexus back in June, 2023, I was only bi-amping while still running the audio signal through the passive xover in my speakers for the mid range and tweeter. I recently purchased a third amp and modified my Eminent Tech 8bs to bypass the passive xovers altogether. For subs, I’m using a pair of dipoles. I do not use the 8b subs at all, definitely the weak link with these speakers.

Just today I got the system up and running. First impression is I’m amazed. This is the future of audio. The DSP Nexus in combination with the 8bs and high quality subs is pretty darn stunning.

@jaytor 

Great system, and sorry I'm arriving 2 weeks late!

I also do DIY, dipole, AMT tweeter driven by SET, although have 4 sealed DIY Rythmik subs. And use DSP for digital xo and room correction. I use Acourate software for this, and believe Audiolense is also great. A great write-up by Mitch Barnett can be found at Computeraudiophile from years ago - there is a basic walk-through and an active xo walk-through, where he explains the setup and process, and results. He late wrote an ebook, and did an article on Audiolense I think.

My setup is an "audiophile" computer running Roon, convolving filters made with Acourate that account for linear phase xo and digital room correction (and time-alignment, etc), that outputs to a 8 channel DAC (Lynx Hilo) thru USB, that drives 8 amps directly connected to the drivers. Hilo has DAC and ADC capability, which is important to take the measurements to create the filters to be convolved. This is sort of the DIY approach to what boxes like DEQX do. One ADC/DAC I've had my eye on is Merging Hapi, but haven't gone there yet.

As I started venturing into active xo, I first used the Rythmik xo to relief my tube amp from reproducing sub frequencies. This was good, but going active with Hilo/Acourate far surpassed this solution. To me there is no going back. Given your DIY skills and willingness to dedicate time to get it done, I suspect you'll be on a similar path.

Please keep us posted on your journey!

Minidsp Flex's are way better than you think.  You can use two 4 channel out ones ($500 each) in parallel and get 8 analog channels out with both digital and analog inputs.....I  read that these newer units are using Burr Brown PCM1795 DAC chips....not confirmed.  These units run at 32 bit 96K unless you are using the option Dirac thang which operates at 48k.  The new Flex units run at higher bits and have better distortion, etc. than most of the older stuff.  Built in digital volume control and remote with several presets.  Here are some measurements:

https://www.audiosciencereview.com/forum/index.php?threads/minidsp-flex-review-audio-dsp.30804/

I spent a while researching the Trinnov Nova, but decided to go with the Danville dspNexus. 

The Nova would have cost $5500 ($3500 + two additional 2-channel licenses at $1000 each) compared to $3000 for the dspNexus. But even at the same cost, the Nova is really designed as a pro-audio studio calibration system and is not as well suited for my purposes. For example, it does not have an IR remote, and it doesn't have an easy way to control the output volume. The user interface is fairly slick, but is designed to be fairly automated and doesn't appear to have the programming flexibility of the Audio Weaver software used by the Danville.

The dspNexus has both a stepped attenuator (in 3db steps) to set the max output level, and digital based volume control that can be controlled by an included IR remote. It is very modular allowing the DSP and DACs to be easily upgraded. Al Clark indicated that they are working on upgrades for both, and a new (significantly more powerful) DSP board will be provided for free later this year. 

Also, the dspNexus will ship on Monday, while the Trinnov is currently backordered. 

@erik_squires LOL

@jaytor I believe the Trinnov Nova retails for about $3500 for 2 channels and $500 for each additional pair of channels so $4500 for a 6 channel unit.

I view digital crossovers, room correction, time domain corrections and amplitude domain correction all very related. Trinnov units do more than simple digital crossovers. It maps the room and builds filters so Your drivers work in Your room.

BTW, Nice system. It's somewhat akin to our Apollo system being light membrane technology and OB. Cool 300B amps. We typically use 300B mono amps in our showroom. During development we have gone through passive crossovers, analog active crossovers and several DSP systems. IMHO, you are on the right path going with digital crossovers. High quality room correction is something to seriously consider, IMO.

@jaytor --

Kudos for venturing into outboard active configuration (it’s what I do myself). It will be interesting to learn of your findings (if you’re willing to share them) compared to your passively configured main speaker system once you’ve initiated the process and has gotten your head around it with actual impressions of the sonics to follow. Cool main setup, btw. - I’m sure it’s very capable.

My biggest reservations are 1) giving up my Denafrips Terminator+ DAC and nice-quality DIY preamp, and 2) using the DAC’s digital volume adjustments.

Your main priority is using the digital input on the dspNexus, I see, which - as a simplified system, with all that may entail - is what prompts above quoted concerns of yours. While on paper/in theory I understand worrying about introducing extra conversion processes, as an outset at least I wouldn’t be too concerned about them. What do you gain by avoiding them, while at the same time losing out on your Denafrips DAC and separate preamp?

Seems to me some effort of yours has been invested in sonically "shaping" the sound of your setup with these very components, so why not start with them remaining in the chain and instead use the analogue inputs of the dspNexus directly from your preamp? Yes, you’ll have an extra A/D conversion (the effect of which is blown out of proportion, if you ask me, if it’s even noticeable), but this way I’d argue you have better grounds for comparing the sound from the passive speaker config. and the one introduced with the active ditto on the basis of this alone.

There’s also the option of a software-based DSP preceding the DAC, which then necessitates a separate DAC channel for each of the amp dittos - like from the Okto Research dac8 PRO. This to some can bring into question whether a theoretical lack of overall signal symmetry of the DAC channels may have an effect on perceived sonics vs. only using 2 DAC channels that are then branched out on the output side of the digital crossover. I gather it’s in the splitting hairs department, but I have no experience to speak of here - not via my own setup, that is.

Regarding setting filter values good advice has been provided already. I’d experiment with steeper slopes than 4. order - myself I use 36dB/octave L-R throughout (BW slopestyle on the subs HP). Does the dspNexus use linear phase filters (FIR)? If not it’s not an issue, as I see it. In any case I gather you’ll be able to get even better integration between you sub and main towers fiddling with delay, although you may come about this by simply moving them back and forth to each other the way you've positioned them. With my speakers we used nearfield measurements of the MF/HF horns to set precise filter notches and make a light peak suppression. On top of that the real work was getting the delay right - not least between the woofer section of my main speakers and the tapped horn subs, where the "origin" of the front wave is somewhere inside the horn path, and not simply on the cone backside exposed at the mouth. With the tools at your disposal already I’d say you have a very good outset for interesting results.

@arion - Thank you for bringing this to my attention. The Trinnov Nova looks like an interesting unit, but is considerably more expensive than the Danville dspNexus, particularly for the six channels that I need. It also seems to be focused more on room correction than on crossover use, but I assume it can do both. 

Since I'm not sure if going active is my long term plan, I don't want to spend too much to give it a try. If I like the results, I might consider spending more in the future. 

Oh, now that I know @arion is a Trinnov vendor I'll be sure to butter them up whenever I can. 

😂

BTW, the Trinnov units all come with excellent room correction and use a very unique mic. They do an excellent job in the amplitude domain and more importantly in the time domain.

I suggest you investigate the new Trinnov Nova. Is has up to 6 channels. We supply a Trinnov DSP unit with all our Apollo and Athena series speaker systems. They are excellent and very easy to set up. Check out some of the show reports about our systems. You will find them to be very positive. If the Trinnov DSP wasn’t good our systems wouldn’t sound good. We don’t sell just processors so I’m not trying to sell you anything.

Hi OP!

If your crossover is that simple then DSP should be fairly straightforward.  The only thing I could imagine you may wish to consider is driver polarity, delay and slopes.

The advantages of DSP include being able to set delay times digitally, and while it may not make a difference for your upper drivers, you can get fine grained phase and polarity matching with your subwoofers.  Also, 4th order slopes may give you better horizontal responses and power handling.

@jwr159 - I talked to Al Clark on the phone for close to 40 minutes today (he likes to talk) and I'm going to go ahead and order the dspNexus from him. 

Hard to say how complicated this will be without knowing your system, and what the passive crossover does.

@erik_squires - My system is described on my Main System virtual system page, but to elaborate a bit more - my speaker system is a dipole line array with Bohlender-Graebener NEO10 midrange drivers, and NEO3 tweeters, with dipole woofer towers, each using four 12" servo woofers. 

The BG drivers have a fairly smooth response, so the crossover is fairly simple - 12 db/octave LC filter for both NEO10 and NEO3 driver arrays, with an additional LC notch filter on the NEO10s. The woofers are driven by Rythmik plate amps that have their own active crossover (along with a simple parametric equalizer). 

I don't think it will be hard to set up a DSP to do the same thing, although I'd probably use a 4th order crossover since it's fairly simple to do with DSP and makes it easier to get a smooth phase response. 

The NEO line arrays are fairly efficient at about 98db/w. I'm currently driving them (using the passive crossover) with a 300B PSET amp (monoblocks). With an active crossover, I'd probably continue to use this amp on the NEO10s and build another SET amp for the tweeters. 

I may be able to bypass the crossover on the Rythmik amps when using the dsp. If not, I'll set it to it's maximum crossover frequency and work around it with the dsp.

I currently use REW with a calibrated UMIK-1 mike. I suspect this will be adequate to set up the active crossover, but I'm open to suggestions. 

My system is digital only, and my speakers are 3-way, so it’s not particularly complicated.

Hard to say how complicated this will be without knowing your system, and what the passive crossover does.

While setting a DSP crossover is as simple as moving sliders around on your screen, entering the right values is more complicated.

A crossover in a passive speaker is almost never "just" high and low pass slopes, but I suppose crossovers for high efficiency speakers are often close to that.

The real question is how well you are able to analyze the current behavior of your speakers and to make the appropriate adjustments.

Why talk to Danny Ritchie or anyone else. Just call Al Clark at Danville. Interesting guy, claims to have design more DSP boards than anyone on the planet, but has an analog background as well. And Al is generous with his time as well. I purchased the Nexus DSP and love it.  I’ve never tried the miniDSP products, but can’t imagine they are as good. The brand new DEQX boxes are probably 2X or 3X the cost.

Ah, I see. I haven't kept up with new models from Rythmik. A pair of HX800's makes a lot more sense than four A370's!

@bdp24 - I don’t think Danny is a fan of active crossovers, but partly because a lot of his excellent skills in passive crossover design. But Richard Hollis from Hollis Audio Labs has had some good success (according to others) using DSP crossovers and a system very similar to mine. 
 

I’m using the Rythmik HX800 amps for my woofer towers. Each of the amps has two outputs, each of which drive a pair of 16 ohm speakers in parallel. With this load, they’re rated at 400w for each of the two outputs. 

@jaytor: Give Danny Richie a call. He’s been involved in some serious investigations into high transparency DSP projects and products. He found the sound of the digital crossover in the Linkwitz LX521 loudspeaker seriously lacking, but there are much better digital systems than the ones Linkwitz used.

Say, I see you now have a pair of 2-woofer H-frames on each side, replacing your precious pair of 3-woofer frames (one per side). Do you have a Rythmik plate amp for each 2-woofer H-frame, for a total of 4 plate amps?

I made the 2-woofer version of the OB Subs, and bought the 8 ohm version of the woofers from Danny, rather than the 16 ohm. That allows the plate amp to feed the woofers their maximum possible output. Being solid state, the amp producers more power into 4 ohms than into 8, of course. And two of the 8 ohm woofers present an approximately 4 ohm load to the amp, while a pair of the 16 ohm an 8 ohm load.