The EA DSPNexus with NX-Treme and Servo Subs Integration Thread

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  • Jr. Member
  • Posts: 22
Hi All,
Good News Everybody! My DSPNexus shipped today!   :thumb:
Thank you HAL!  :notworthy:

So lets get this “build” thread started.  My goal is to not only share my journey, but to provide a reference point for members of the community interested in taking their path to audio transcendence down the road of DSP and active crossovers.

This thread starter has tons of detail, but I think it all matters in the greater scheme of things. 

I believe I am typical of audiophiles on a budget who do their best with what they have to develop their systems over time.  Over the past 25 years, I focused on the best bang for my buck with each improvement.

Except for the speakers, this is the biggest upgrade I have ever made.

I am not an audio engineer.  Not even close.  I learned my way around REW through a “monkey see monkey do” approach.  I also built my chip amp using the same approach.  So, bear that in mind when I start asking stupid questions.   :scratch:

My interest in DSP and active crossovers started pre-covid when I was watching HAL’s thread on adding his monolithic 6x12” open baffle servo subs to his line arrays. 

About a month ago, I was chatting with @captainhemo when he suggested looking into the DSPNexus and going active.  I don’t know what it is about that guy, but he rivals the skills of Satan when it comes to sowing the seeds of audio temptation.  I have made more changes to my system based on conversations with him.  That guy kills me!

So I reached out to HAL with all sorts of questions.  His thorough answers showed his passion for helping others improve their systems which ultimately brought me here, taking it all to the "next level" on a system that is already considered at the "next level" by many.  This hobby is brutal.   :icon_twisted:

At this point it is in my best interest to add that my wife is a saint! :angel:  I know that because when I show folks my system, the first thing they say, without fail, is “Wow!!  Your wife is a saint!”.  I consider myself very lucky she puts up with my obsession.

As Is
I am running GR-Research NX-Tremes with what I affectionately call the “Don Bombs Crossover Network”.  This is the first NX-Treme kit ever built by @captainhemo and his buddy Don.  Don sparred no expense in doing it right and worked with Danny Richie to take the crossovers to another level using the best copper he could find.  You can see some pics of that here:  So active crossovers have their work cut out for them in this upgrade.   :D

The speakers are bi-amped using a McIntosh MC7270 for the mids/tweeter (Nordost Red Dawn speaker cables) and a Folsom DIY 7293 chip amp driving the 8 woofers on each channel (DH Labs T14 speaker cables).

I am supplementing the bass with the 3x12” open baffle subs.

Pre-amp is a Rowen PA1.  This is a Swiss built piece of kit that is very transparent.  As it is purist, that means it is all manual and no remote control.

My cables are mixed.  The interconnects are primarily DH Labs White Lightening and Bluejeans cable to the subs. 

My digital music is controlled by Roon (Tidal, Qobuz and CDs burned to my NAS).  I am using convolution filters I built using REW for room correction (shout out to Obsessive Compulsive Audiophile for his videos on how to do this).

A Raspberry PI plays the roll of a Roon bridge to my Topping D90MQA DAC.

Everything that runs through my projector (Nvidia Shield, PS5, Sony 4k Bluray) is fed to the system via an optical SPDIF cable to the topping DAC

Old school pioneer turntable with a Denon 103 cartridge.

I alternate between a Parks audio budgie (tube phonostage) and Puffin (very fun ss phonostage).

Damn near a square (20 X 21 X 7.75).  While the figure 8 sound wave pattern of open baffle helps with that I need to beat it into submission with quite a bit of diffusion and a makeshift bass trap in the back corner.

To Be
The arrival of the DSPNexus in will change things up a bit.  First, it is the active crossover for the subs, the woofers, the mids and the tweeters.  It will also take on some of the pre-amp functions, in particular volume control for the whole system.

I will add a topping “PA5 II plus” amp due to its solid review on audio science, and more importantly its price.  My planned assignment of amps is as follows:
  • Subs:   A370 PEQ servo amp
  • Woofers   Folsom 7293 - this amp provides a very full sound and incredible bass
  • Mids   MC7270 – it handles vocals very well
  • Tweeters   PA5 II Plus – considering how clean it is supposed to be I felt this would be best
I will be using 3 different speaker cables based on my perception of their strengths:
  • Nordost Red Dawn for the tweeters
  • DH Labs T14s for the mids (these are surprisingly very close to the Nordost in sound)
  • BlueJeans 10 gauge for the woofers.

Expectations  :rock:
@HAL, since you have my money already, please correct me if my expectations are misguided.

In the spirit of improved sound quality, I am trying to solve for the following:
  • Single volume control for the whole system with a remote.  This is important as that is how I got my loving and supporting wife (ok… she is more tolerant than supporting 😊) on board with the project.  I personally will get off my fat ass to adjust things for sound quality.  That is why my Schitt Freya plus preamp with remote is on a shelf and not in my system right now.
  • Room Correction for all sources, including analog.  Currently I just have Roon
  • Move more of the bass to the subs.  The speakers run pretty low in my room, but I think I want the subs to handle more of the bass, maybe up to 60 or 80hz.  I am open to suggestions here (including reasons not to mess with it).
  • Easy and affordable upgrade path.  Not just from the DSPNexus, but I am thinking I may eventually want to connect directly to the speakers from the amplifiers without speaker cable runs.  If the Topping amp cuts the mustard, I may buy 3 more of them to accomplish this
  • Higher quality DAC for my BluRay player.  I love watching concert videos (Talking Heads “Stop Making Sense” is often asked for when I have folks over), so I am hoping a direct connection into the DSPNexus will do that.  But will need to see what inputs I have available when all is said and done.
  • Ability to learn more about sound engineering.  I am anything but a sound engineer, but this piece of kit should give me enough to be dangerous.
  • The versatility of the DSPNexus will be attacking the barriers to audio Valhalla from many different angles, so this being fun is an expectation.
  • As an early adopter, I hope to discover challenges and opportunities that lead to product improvements and solutions.
  • And of course, I am looking forward to hearing the improvements afforded by active cross overs.  With the “Don Bombs” being the pinnacle of passive crossovers for the NXTremes, this should be interesting.

Constraints/Compromises to the implementation :duh:

As I am not running amps directly to the speakers at this point so I need to couple the speaker wires to the speaker cables.  I will be using cheap things I found on Amazon.  I do not want to invest in decent couplers when the goal is to eventually connect the amplifiers directly to the speakers without speaker cable.  If this proves problematic, I may re-purpose the tube connectors on the passive crossovers to improve the signal path.  If get flamed hard enough I might even solder the connections.

My current two amplifiers are not balanced, so will be using Neutrik XLR to RCA adapters.  This should not be an issue, but once again, this is not the most optimal.

Thread Starter Conclusion
Please feel free to question anything I said.

I will edit this starter as appropriate for grammar and clarity

So with that said, lets all patiently wait for the FedEx truck.



The dspNexus 2x8 should do what you are looking for even without the first upgrade that will be coming for the EA units that will be shipped when done.  That is the new ADSP21569 dspBlok from Danville Signal that will add a lot of throughput and new capabilities.  One of which is having both long FIR filters that process at the same time as the DSP path.  I have a few customers that already will be able to take advantage of the new capabilities. 

The new hardware and firmware are in testing, so hopefully it will be soon.  The 4 DAC's, DSP module and ADC's are all plug-ins, so not a hard thing to upgrade.

REW is being used by most customers now to add room correction below 300Hz.  Simply measure the room with REW and a XLR phantom powered mic input on the front.  Once the measurements are complete, REW can calculate the room response PEQ's to flatten the response.  Those can then be entered into the Audio Weaver block diagram for each channel for trials.  Once you are happy, then the block diagram can be compiled and downloaded to the dspNexus 2x8 to use every time it is started. 

If time delays are needed to compensate for the NX-Tremes position relative to the servo subs, that can be input to the block diagram.  That makes a big difference even for systems with Line Force speakers and 3x12 servo subs.  :)

The remote control and battery are in the box.  Once you have your Audio Weaver software downloaded, I will help with the installation.  That is part of the documentation that Danville has not finished, but after over 10 years of using the Audio Weaver software, have a pretty good idea of what is needed to get you up and running.

The unit should be there this week, so the fun will start soon.  :)


  • Jr. Member
  • Posts: 22
Good News Everyone!

My DSPNexus arrived and it is time to play. 

I will do two posts at a time.  The first has the subjective stuff and experiences, the second is procedural stuff to easily reference for those taking this path.

I do not have the exact crossover specs for my speakers yet and am waiting for an Audiomatica Clio to measure those values, so today was basic setup and testing to make sure it is all in order.

The first step is using AudioWeaver, to transform the dspNexus into a DAC .  It takes a while for configurations to be applied to the Nexus, so testing it as a DAC first makes a lot of sense.

All the outputs on my unit tested out fine.  I did have issues with the headphone jack, but the ¼” adapter is looking more suspect than the Nexus.

Impressions, Challenges and Considerations

Inputs and Outputs
Besides the obvious 8 XLR channels out, it also has an SPDIF in and out (Both are RCA type coax connections), Analogue balanced in, and a USB type A in.  On the front of the unit there is a head phone jack and an XLR in for a phantom powered mic.
The menu system is pretty easy to navigate, I do wish it was customizable.  For example; every source has an inverted version of that source (USB and USB Inverted, SPDIF and SPDIF Inverted, etc…).  This not something I will likely be using, so customization would be nice, but I am being nit picky and I am sure the development team has enough on their plate right now.

The DAC sounds very nice and clean.  It is a bit early in the process, but I feel it is rendering quite a few more details than my Topping (which is one helluva DAC for the money).  So, we are off to a great start.

I did not listen too critically but noticed more breath in the horns with more details in the strums and plucks of acoustic instruments.  It just sounded more real to me, like the characteristics of the instruments was more dialed in. Considering Danville uses only the best DAC chips on the market, I would hope this to be the case.
The analog input is transparent, there was no veil or colorization that I noticed right off the bat, which is exactly what I am looking for.  I was most happy to being able to adjust the volume of evenings entertainment from a remote (no remote with current pre-amp).  It helped ease my wife’s pain of seeing cables strewn across the living room.

It’s Called “Early Adopter” for a Reason Folks
We did have a few issues to investigate. 

In all reality, these are all minor considering how much configuration is done by the end user.  No major headaches that a little aspirin won’t sort out.

S/PDIF Output
The first challenge we hit was trying to use the S/PDIF out.  We were not successful in getting that to work with the Topping.  Not super high on my priority list, but S/PDIF out is a pretty cool feature I can easily find uses for.  So will keep this on my radar. 

XMOS Drivers
Apparently, there are known issues with the way Roon communicates with the XMOS drivers on Linux based bridges/endpoints. 

After about an hour of listening I noticed distortion that started as clicks and pops and quickly morphed into fully distorted music.  I adjusted the sampling conversion settings in two places on Roon - The device specific settings and from the “Muse” window.  This seems to have solved it but will only know with time.

To be sure the distortion was not caused by the Nexus, I hooked up my old Linn Majik without touching the Roon settings to see if I had the same problem.  It has been playing happily for a couple hours as I type this up.  BTW, this DAC and the Linn are a great combination.  The Linn is an older model and had a habit of needing a hard reboot from time to time.  This was quite annoying so it will not find a permanent place in my main setup.

There is Always a Solution
Due to the potential of those Linux drivers going flakey I decided to give one of HALs windows based servers a try.  Will hear how that sounds and report back.

Crash and Burn
I did manage to crash the Nexus at one point requiring me to reload the DAC configuration from AW.  HAL is working with Danville to see what is up with that.  Not too worried at this point.  This early in the product lifecycle, there are bound to be a few bugs that will result in such behavior.

Honestly, I think I would have been disappointed if I did not manage to crash it once. Like I was losing my touch or something.   :lol:

So far so good and no major regrets.  The real fun will begin once we start with building the active crossover network and getting the timing right.  I have high hopes for this unit and so far I do not see any reason it may disappoint me.

I will post some pics eventually.


  • Jr. Member
  • Posts: 22
@HAL - Keep me honest here and please let me know if I missed or misrepresented anything.

Procedural Stuff
So here how you setup the nexus for the first time to test that it is working as expected.

Getting the Needed Software
First things first.  Get AudioWeaver up and running. 

DSP Concepts (AudioWeaver) Account
Open up an account with DSP Concepts (  This is required to get the software licensing to work.

Word of advice: when creating your account, do not use the fancy automatic browser secure password generator.  Pick something you can remember or you will find yourself fumbling around your browser settings trying to find the password so you can cut an paste it in.

The Nexus unit comes with a license which is sourced and managed by Danville.  It is updated yearly, so don’t worry when it starts warning you it is soon to expire as it will work the day after. 

The Software Package
Danville has their own version of AW which they send you before the Nexus arrives with an email containing a DropBox link.  It’s about 700MB of Zip file to download.  Having a speedy internet connection is nice.

HAL also sends pre-written code to turn the Nexus into a DAC for the first part of the installation and testing.

Install the XMOS Drivers
These are supplied in the zip folder and it is pretty straight forward.

The Trickery When Installing AudioWeaver
Everything to get the software up and running is in that zip file but there is a little trickery you need to do after the installation. 

Post Install
1.   Create a folder called Archive in C:\DSP Concepts\AWE Designer 8.C.1.3.B Standard\Bin\ (This path is the default path for the windows installation.)
2.   Copy these files from C:\DSP Concepts\AWE Designer 8.C.1.3.B Standard\Bin\ to your newly created Archive folder.
a.   AWE_Server.exe
b.   AWE_Server.ini
c.   FrameDll.dll

Now that those are backed up, we overwrite them with the files of the same name found in the zip file provided by Danville.

You now have a Nexus friendly version of AudioWeaver.

Firing AudioWeaver Up
Use the shortcut the installation put on your desktop.

Login In
When starting AudioWeaver it asks for your “DSP Concepts” login.

Two Windows for the Price of One
after you enter your login in details, two AudioWeavers windows open up. 

One is called AWE Server.  This has connectivity related info.  The other is AWE designer and it looks a lot like Visio.

Connect The Nexus
The nexus comes with a USB B to A cable (the same that most old school printers have) if you do not already have one. 

Connect the nexus to your computer and then turn the nexus on.

Open the “Server” window and from the top menu select “Target” and then “Change Connection”.  This connection entry is defaulted to “Native”, click the down arrow in the box and select “DanvilleXMOS” and then close.

You will see under the “Output” tab a heading of “Target Information” with the dspNexus as the name.  If you don’t, something went horribly wrong.  Let HAL know.

Import Files
Import the file Rich sent you called dspNexus_AWD_multifeature-DAC8-2xxxxxx.awd (the “x”s are the version number).

Load the DAC functionality to the dspNexus
There is a blue icon that looks like a “Play” button.  Click that and AudioWeaver will do its thing for a minute to compile and upload this DAC functionality to your Nexus.

Test by Playing Some Tunes

Safety First - Turn the volume down on everything you can turn your volume down on. 
Do this with every new thing you connect to the Nexus.  No one wants surprises in this arena. 

Now your Nexus is a very nice functioning DAC that we can start testing. 

Keep your USB connection to your computer.

Test the Ports (Plug Stuff In)
The easiest way to accomplish the first test is to plug in some headphones, hit the youtubes or your favorite music player and listen.

More tests and if you do not have headphones:
My headphones only played one channel.  However, I was using a potentially janky and very untested adapter, so I will not blame the Nexus yet.  I sprung plan B into action (which is the next step anyway) and plugged the outputs from the number 4 and 5 XLR outs into my pre-amp and tested that way.  It worked!

The Nexus has XLR outs 1 - 4 assigned to the right channel and outs 5 - 8 assigned to the left.  Test all channels to make sure it is good to go.

Hooking Up Your Sources

BEFORE DISCONNECTING THE COMPUTER FROM THE DSPNEXUS – Go to the server window and click the disconnect check box in the lower right corner.  This finalizes the connection properly and makes sure your configuration stays on the Nexus.  Think of it as safely removing a drive, but Danville really means it!

Conclusion of Initial Install and Testing
At this point I was for the most part done using my computer so I connected my streamer via the USB port.  I use a Raspberry PI with DietPi and the Roon bridge.  I also have an older Linn Majik that I added to the system for reasons stated in the post above.

I hooked my current pre-amp up through the analog section to watch a show and test how that works. 

I may use the phono-stage capability in the future but still trying to figure out how to get all my sources routed through dspNexus effectively.

There You Go!
You are now able to play music through the DAC, if not, I suggest calling HAL at this point.

I believe the next procedural post is adding in the active crossover programming and getting the timing right.   :popcorn:
« Last Edit: 15 Sep 2023, 08:46 pm by Kazoom »


Great write-up of the EA dspNexus 2x8 system setup!

Only comment is that the Right channels are 1-4 and the Left channels are 5-8.  That should put the violins on the left and double basses on the right.  My typical test to see if I did connect it correctly.

Getting the package ready with the HAL MS-6 and CLIO Pocket to make the crossover measurements to get the NX-Tremes system running as a 3-way XO with your servo subs.  The MS-6 will run the CLIO Pocket for measurements and can act as the ROON Bridge from the ROON Core system via Ethernet or WiFi.  I would use a 5G WiFi connection for music transfer if not hardwired Ethernet.

If you measure the distances from the NX-Tremes to your chair and the center baffle of the servo subs to your chair and see what the difference is, that becomes the time delay needed to time align the speakers.  A great thing to do with the DSP crossover.  Each channel can have a different time delay if needed.  Have done that for one customer that has his woofers and subs running to make flat response with REW measurements and the PEQ's filter data it generates.



  • Jr. Member
  • Posts: 22
Fixed the Left/Right in the post.  I rolled the dice on that one.   :lol:

I did test by killing the left channel in Roon and sure enough, music only came from the right.  So got it half correct.   8)

Dialing this in is going to be a ton of fun.


Glad to hear you are making progress. I've had my dspNexus for a couple of weeks. It took a little finagling to get the USB to work from my computer. I adjusting the timeouts as explained in the dropbox installation doc, but it was still not working reliably to generate files on a big design like the three-way crossover.

I had left my dspNexus powered on and connected to the computer over night and the next day, it worked fine with no hickups. So my guess is that the timing changed subtly after it had warmed up for a while.

I bought an inexpensive active speaker (Rockville RPG8) to use for testing. This has a volume control so I can turn everything down when I'm playing with filters, and I don't have to be concerned with damaging my expensive speakers.

I am hoping to find some time this weekend to dig into this to see how much I can get down in setting up the full crossover. I am planning to use this with my Line Forces and OB subs, so I'll be bypassing the passive crossover completely.

I'm thinking of starting with 24db/octave crossovers at 180Hz and 1800Hz. Rich - do you think this sounds reasonable?

Kazoom - are you using IIR or FIR filters for your crossovers and PEQ?


The crossover points sound reasonable for the BG NEO10 and GRNEO3 drivers.  Been 8 years since I measured the first pair at Danny's with the dspMusik 2x8.

I typically use 8th order XO's with planar drivers.  Works well with the BG NEO10/GRNEO3 combination in the Super Mini's using the dspNexus 2x8. 

Also add the time delay offset if the line array and servo subs are offset in distance to the listening position.  That was a good sounding change when we did it for imaging.

If you do use FIR filters there is a change to the AWE Server INI file that eliminates a weird USB related signal that is being worked on for FIR's.  The change is being checked in a new system on Sunday, so will see if that solves the issue until the fix is implemented.



  • Jr. Member
  • Posts: 22

I have not done anything with filters or building the cross overs yet, nor do I totally get the technical ins and outs between the two and why FIR is considered better for audio and IIR for video but yet both are used in both applications (engineers are so confusing).  My suspicion is I will be using FIR but I am no audio engineer, and will take HALs guidance to get this up and running. 

Once I am stable, happy and my configuration is backed up and locked away offsite I will likely venture into playing around with pretty much anything someone thinks should be tried.  My love for hi-fidelity is only surpassed by my love to mess with stuff to try and find a better way.  From your posts on the various sites we frequent, I think you are in the same camp.  😀

Part of the purpose of my thread is to get ideas to try from folks way more versed in the technical aspects.

BTW, I would love to hear your set up the next time I am in the Portland area (I believe you are around there), you have a ton of experience and ideas I would love to steal...  I mean listen to.   :green:



BTW, I would love to hear your set up the next time I am in the Portland area (I believe you are around there), you have a ton of experience and ideas I would love to steal...  I mean listen to.   :green:

I'd be happy to have a listening session when you are down in Portland. I assume you live in the Seattle area? Before moving to Portland, I lived in Sammamish, Bellevue and Redmond.

I played around with the dspNexus a bit today and I'm a bit confused. The sample 3-way crossover design seems to be working except for output 1. Instead of the high-pass output from the right channel, it seems to be outputting a full-range single from the left channel. All the other outputs are working as I'd expect. I don't see anything obvious in the Audio Weaver design, but it is pretty complicated and I'm FAR from being competent with this program at this point. I sent an email to Danville to get some assistance.


Post a screen capture of the crossover block diagram and I might spot something for some feedback.

Just need the crossover section not the upper level controls block diagram.


Post a screen capture of the crossover block diagram and I might spot something for some feedback.

Just need the crossover section not the upper level controls block diagram.

Thanks. Here it is.

Low distortion

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  • Posts: 39
Can someone explain to me what this product has over the Minidsp Flex 8?

 Minidsp shares the measurements taken with an APx555, it’s dead easy to use, and only costs $599. Has anyone compared the 2 subjectively? And do we know the objective measurements of the Nexus?


  • Jr. Member
  • Posts: 157
Already asked
« Last Edit: 20 Sep 2023, 02:08 am by david45 »


Can someone explain to me what this product has over the Minidsp Flex 8? (

Minidsp shares the measurements taken with an APx555, it’s dead easy to use, and only costs $599. Has anyone compared the 2 subjectively? And do we know the objective measurements of the Nexus?

Just asking out of curiosity

At this point, the dspNexus uses essentially the same DSP as the miniDSP you linked, but the dspNexus uses a modular architecture and Danville has promised a free upgrade to the 21569 which is way faster, particularly for FIR filters.

The Audio Weaver software is certainly more complicated than the software Dirac includes, but also WAY more flexible, so kind of depends on what you are after.

I would expect the dspNexus DACs and analog hardware to be higher quality. The power supply is certainly higher quality, and it has balanced outputs with programmable analog gain. With the miniDSP, you have to give up more of the DACs dynamic range for level adjustments between outputs.

The dspNexus includes a high quality analog input if you want to use analog sources, and a powered microphone input allowing the use of a higher quality mic for measurement and calibration.

The miniDSP is a great value if you want something simple though.

Rich may be able to chime in with more detail.


Let me see if I can help with the dspNexus 2x8 system understanding of differences.

The EA dspNexus 2x8 uses an ADSP21469 UAC2 dspBlock as its DSP engine, this is to be updated soon to the ADSP21569 UAC2 dspBlok that will be more capable with the MAC and FIR processing of long FIR's done simultaneously with more memory available.  There are customers already building FIR based crossovers that require the ADSP21569, so those will be upgraded for free for the Early Adopter units.  The final version dspNexus 2x8's will the ADSP21569 dspBlok.

The system uses an AKM AK5578 ADC and four AKM AK4493 DAC's in the standard 2x8 configuration all balanced connections.  It has remote selectable ADC, S/PDIF and USB2 inputs, where the ADC's can be used for line input or RIAA phono input via a balanced connection.  Both the ADC and DAC's have level gain controls and also volume controls on the DAC's via the remote. 

The system it totally modular, so the ADC, DAC's and DSP are upgradeable as time goes on.  There are already plans to offer an AKM AK4499EX version of the DAC boards.  A prototype of the AK4499EQ boards was done at LSAF2023 with very high praise from both consumers and companies who attended to listen.

The AKM AK4493 DAC's are 32bit so that they have extended range for digital volume control.  The AK4499EX DAC is the next generation DAC that will be an upgrade.  I have replace a Pass Labs Aleph P and Ono with the dspNexus 2x8 as the front end.

Audio Weaver is a considerably powerful programming system for the dspNexus DSP.  Every channel is independent and programmed as needed.  This way Filters and PEQ's can be put on the channels where they are needed, not just a given number per channel. 

The system as pointed out has an ASIO USB driver that works with PC and MAC, and will work with Linux if the person setting up the system knows how to go into their audio system and select the correct output. 

There is no RPi4 in the system, and only a Windows PC can be used to program the dspNexus.  Once the program is the way the customer wants, it can be flashed to the DSP and will always run on power-up.  It can also be changed at anytime with Audio Weaver for updates to the block diagram.

Audio Weaver has high precision audio filters available that have been compared to the typical stock DSP filter topologies that have lower distortion.  That is a propriatery capability of the system.  The Linkwitz-Riley multiway crossovers are one example of a high precision filter.

The system allows time delays per channel in either sample, time or distance options to correct for things like main and subwoofers being placed at varying spacing to deal with room acoustics. 

There is a headphone output that disconnects the speaker feeds automatically when plugged in.  It becomes a full bandwidth DAC channel and the headphone amp will driver planars like Audeze LCD-MX4's. 

Next to the headphone output is a phantom powered XLR mic input for acoustic measurements that is compatible with Room EQ Wizard.  This is being used by customers to dial in the room response below 300Hz for multiple woofer/sub configurations. 

This has also been A/B'd in a DEQX Express 2x6 system with a DCX2496 running the subs and the customer chose the dspNexus 2x8 as it sounded better.  He now uses it with REW for his acoustic measurements and PEQ settings with his ACO measurement mic and preamp. 

I have suggested that the dspNexus 2x8 be tested with one of the Audio Precision systems that Danville has access to via another company.  They also know about ASR doing measurements.  That is their decision.

If you have any other questions, please let me know.


  • Jr. Member
  • Posts: 157
At this point, the dspNexus uses essentially the same DSP as the miniDSP you linked, but the dspNexus uses a modular architecture and Danville has promised a free upgrade to the 21569 which is way faster, particularly for FIR filters.

The Audio Weaver software is certainly more complicated than the software Dirac includes, but also WAY more flexible, so kind of depends on what you are after.

I would expect the dspNexus DACs and analog hardware to be higher quality. The power supply is certainly higher quality, and it has balanced outputs with programmable analog gain. With the miniDSP, you have to give up more of the DACs dynamic range for level adjustments between outputs.

The dspNexus includes a high quality analog input if you want to use analog sources, and a powered microphone input allowing the use of a higher quality mic for measurement and calibration.

The miniDSP is a great value if you want something simple though.

Rich may be able to chime in with more detail.

Thank you for the reply, Jaytor.

You set the bar pretty high with your passive crossovers and at the end of the day, what matters is how it sounds in your system. I’m super happy to hear that you can get the free 21569 upgrade. Hopefully it’s a good fit for you when you get it to work and everything is dialed in perfectly :)

For the rest of us average users, measurements, subjective comparisons and ease of use comparisons would go a long ways.


  • Jr. Member
  • Posts: 22
I will post measurements as I pull them.

Clio arrives Tuesday afternoon as well as the HAL music server, so lots to get busy with.  Will likely do a recap of the day on Thursday with the procedures, impressions and next steps.

But first a little house keeping, discoveries and further impressions.

XMOS and RaspberryPI
I let the RasberryPI streamer play over night trying to reproduce the distortion issues I mentioned, and so far I am not able to.  I am quite confident that the Roon settings have solved that issue.

Tried out the Bluetooth connection today. Aside from my car and work headsets, in the last 10 years, I have not used bluetooth for anything audio in the home.  I have always noticed the degradation in the sound and just did not bother. 

I will assume the technology got better since I last tried because The dspNexus did one heck of a job with my Qobuz stream from my phone.  I am most impressed.  Bluetooth is a definitely a viable source.

Menu Customization
You can customize the menu.  (I spoke too early in my last post).  While playing around I found that you can turn off any un-used outputs and they do not show up in your selector. 

Very happy about that.

DAC Impressions after much more listening
I spent quite a bit of time comparing my current DAC to the "dspNexus as a DAC", and to my ears, it is indeed better on tracks were details really count. 

For example, one of my favorite testing tracks: "Sheffield Lab: Drum and Track disc".  The room they play in reached a new level of life I did not hear before.  I also noticed more feeling/realism with the dspNexus.

Another group I like to test with is "The Dead Weather".  Alison Mosshart's vocals can have quite a bit of sibilance that gets vicious on the ears at higher volume levels, but with the dspNexus I could crank "I Can't Hear You" without without feeling pain.  Still sibilant (as I believe was intended), but detailed and clean.

Enough for Now

What really counts is the final product.

It sounds pretty good so far and we still have the crossovers and timing to dial in.  And then we can see what it does with room correction.

Fun times ahead!


  • Jr. Member
  • Posts: 22
Exciting day for me.  Not so exciting for posting but will post anyway to keep the thread alive and do my best to make it at least entertaining.   :D

We used the Clio and HALs MS 6 (learn more here to measure the crossovers.

The MS 6 was preloaded with Audio Weaver and Clio software, so that little box, with only an Atom processor, was pushed to its limits performing streaming, measuring, dspNexus design loading and email client duties.   

Mapping the crossovers was not the easy task we initially thought it would be.

The simply mad genius wizardry that Danny employs in his designs (including inverted drivers) made this procedure trickier than originally anticipated.  I am just saying that you should not let that humble Mr. Good ol' boy, "Hello Everybody, I'm from Texas" charm fool you.  That man's skills are not to be underestimated or trifled with.  I watched him bring my second gen B&W 801s to a whole new level by overhauling the crossover design.  Furthermore I was told by an Emmy winning studio designer that Danny's set up is the best thought out system he has ever heard.  Straight up unsolicited bona fide kudos!

Mad respect!  :green:

Enough kudos, I now digress.

So HAL and I persevered.  We scratched our heads.  We solved a few puzzles (ok, so HAL solved a few puzzles anyway), and got what we needed in relatively short order.  I probably should provide full disclosure; my roll was really just taking pics and putting alligator clips where HAL told me to, but it was a team effort and we prevailed none the less. :)  :whip:

Due to schedules and the work HAL took on to translate the measurements into an AudioWeaver design, the next play day is looking like Friday.

So Friday will bring something for everyone's reading pleasure as they pre-funk the weekend.  :beer:

PS - As I finish this post listening to Edgar Winter's "Frankenstein" on the "dspNexus as a DAC only" with HAL's MS6, I must share that it sounds incredible enough to.... well.... share.   :thumb:


I had a video chat with Emilson from Danville this morning to discuss the problem I was having with output 1. Turns out that the channel 1 output mux was set to test mode so was selecting the wrong signal. Apparently this is the way it defaulted from the factory. To fix this, I had to select the Menu on the remote and then enter the Measurement function. I didn't have to change anything - just hit menu again to go out of measurement which set the output mux to normal operation.

The output is working as I would expect now so I can start working on testing on my speakers.

Emilson also explained how the output level adjustments are done. The gain block in the Audio Weaver design for each channel can be adjusted for fine adjustments, but the Audio Weaver also includes an analog gain block which can be adjusted with the remote. Click on Menu and then use the left and right arrows to find DAC Settings. Selecting this option with the center button shows a sub-menu which allows selection between DAC level, DAC delay, polarity, etc. The level of all the DACs can be adjusted in 3db steps to set the maximum gain for all channels. This is done with the DAC Level - All function.

Selecting the DAC Level 1-8 function allows the level to be set for each channel separately. There is a < or > symbol shown on the display to indicate whether you are selecting the specific DAC channel or the level for that channel. Clicking the center button on the remote selects which way the < or > is pointing (left to change channel, right to change level). In this case, the level can be adjusted in 0.25db steps. Every 3db step is done by the analog gain adjustment hardware, and the smaller steps are done by adjusting the digital levels feeding the DAC using the DSP.

Next step for me is to make some speaker patch cables so I can connect my individual amp channels to the drivers. Once I know that the dspNexus is going to work for me, I'll do something more permanent with higher quality connections. I'm thinking of building an amp for the high frequencies that will sit where the passive crossover currently resides and will include a speaker-level pass-through for the mid frequencies to connect to a  separate external amp. My sub towers which use the Rythmik HX800 amps will connect to the low-frequency channel.

I'm currently thinking of using my 300B SET monoblocks on the mid frequencies (180Hz to 1800Hz) and building a small class A/B amp (probably a composite chip amp) for the upper range.