Ok, I'm still in the process of building this crossover, but have gotten far enough along where I can speak somewhat intelligently about using the dspNexus, so thought I'd add some comments for you all. I should probably preface this by saying that Danville is still in the "Early Adopter" program with these, and that there will eventually be a lot more resources, etc, to take advantage of. So my journey is probably going to be a bit more involved than someone who comes along in a year.
Anyway:
Look and Feel
You can check out the pics on Danville's site to see what this looks like. Those pics are accurate except you'll get labels on the back for all the ports, etc. (I think they used a prototype model for that). Generally feels very sturdy and well built. I haven't opened it up yet, but will in a few months when they ship out one of the upcoming updates and can snap some pics at that time. I guess some people might like a bit more bling, but bling costs and I think Danville is more concerned about performance than looks, which I can appreciate.
My only complaint hardware-wise is that there's no 12v trigger. Everyone who uses this is going to have a bank of amps sitting there, so it would have been great to allow the dspNexus to manage all of that. I've told Danville, and they acknowledged that a lot of the Early Adopters are saying the same thing, and that they're going to figure something out. I know that everything on this is built in a modular fashion, so we'll see what happens with this piece - I'm really hoping a 12v plug-in shows up with a new backplate at some point. Would definitely make life easier.
Installation
Danville sends over a dropbox folder with a bunch of stuff inside. There are a couple movies you watch, and it shows you what to do. There's a custom Audio Weaver version that works with the dspNexus which you install, then configure some files, replace some DLLs, and install a driver. Takes about 30min if you include watching the vids. No issues. I expect at some point they'll just build an installer that does this part, but it's generally easy.
Audio Weaver
This is the program that you use to build your crossovers. This is a full-fledged audio programming language, and is total overkill for doing just a crossover. Danville uses it to control the entire dspNexus unit, all the way from the inputs to the outputs, and they give you full access to all of this. Just to give you an idea:
I can't even fit all of this into one screen. You can go find the DAC blocks and see how they do it, can trace the routing from here to there, etc. It's cool. The part that you need to edit is the crossover region, which I've circled in red. When you click into that, this is what you'll see:
Danville has set up an IIR-based 3-way here, so if that matches what you're trying to do, you'd just need to edit the crossover points, the EQ blocks, and the delays to match your own setup. In my case, I'm going with FIR, so I had to delete the IIR stuff and add FIR blocks instead. Note that if you were building a 4-way, then you'd just change the "signal pass thru/monitor" line to match the others, and use that for your 4th output. In the version above, the signal just passes through to those outputs, so you could route them over to a subwoofer that did its own low pass. Or, alternatively, build your own low pass right there in AW just how you like. Super flexible.
Once you're done with your crossover, Audio Weaver builds it, and sends it over to the dspNexus, which is connected to your computer via USB. My connection is a little wonky, so I can only get it to work once out of every four tries or so without throwing an error, which is irritating. I probably have some connectivity parameter set wrong and have to reach out to Danville. (Early Adopter teething issues I guess.)
FIR filters in Audio Weaver
There are two FIR filters that I found useful. The first is just a generic FIR filter. You specify how many taps, and load up the coefficients. What you'd expect. The other one is a long FIR filter, which is a more efficient version that can do longer filters (like 1000+ taps) more efficiently by breaking it down into smaller chunks. I don't get exactly how it works from a technical perspective, but check out Danville's AudioWeaver vids for an example.
Because there are multiple FIR filters available with differing levels of efficiency, the question of "how many taps can this handle?" is sort of a tricky one to answer, because it all depends. When you flash your build over with AW, you can run it on the dspNexus and get stats back, like how much CPU/memory usage your design consumes. In my case, I'm using almost 4000 taps between 6 drivers (left and right speakers together), which comes out to 68.7% CPU. Here's what that looks like right now for my design:
If I throw a long filter in there for the woofer instead, CPU drops to 53%, so there's definitely headroom here for another driver. Danville's fall update is going to use a 5th generation SHARC processor, which is even faster, so I don't have any worries as far as running out of taps at this point moving forward.
Measurements
Eventually, the dspNexus will be able to take its own measurements. The hardware is already in place to allow for that. However, they're behind on the software end of this, and I've got no idea how any of it will eventually work. I'm wondering if they plan to release this part once they sort out the RaspberryPi supply chain issues. I think the initial idea was to have some built-in room correction options here. Something to look forward to, but not really an option in the moment.
What this means is that you need a way to measure your speakers. In my case, I've got a Pocket Clio, which I'm pretty happy with. You need something though.
Building the filters
Although AW can implement FIR filters (and IIR too), there's no way to construct any of that in there from a graphical/design perspective. It's more to implement the filters that you already have in hand and know that you want to use. So you need to figure out your coefficients elsewhere. I've used some OEM-specific software before which made all of this very easy, but that won't work with Danville's stuff, so I ended up trying out a couple of programs that were new to me.
The first was rePhase, which I first read about via the tutorials over at minidsp. This is a totally free program that is generously offered to the DIY community. I started here, but quickly became frustrated. You can adjust magnitude and phase independently, but you do that via 16 banks of 16-band equalizers. Basically a crap-ton of EQ on each curve, and you can sort of squish it into the shape that you like if you've got the patience to futz with it. I don't know how anyone manages to correct high frequency phase issues with that. I gave it a couple days, but eventually gave up. I think rePhase is designed more for correcting speakers with passive crossovers rather than building a multi-way speaker from the ground up.
I ended up using Eclipse Audio's FIR Designer, which is pretty amazing. It takes a few tries to get the flow down, but once you understand how that software works, you should be able to build filters to do pretty much whatever you want. (The ones I built start off flat in the bass, then drop off by -3db by 20khz.) I'm sort of bummed that I finished those, because I want to fool around with that software more.
Anyway, the downside to FIR Designer is that the multiway version is expensive ($500/year), so this is on the opposite end of the spectrum from rePhase. I didn't want to skimp out on crossovers though, since they're arguably the most important part of the whole design, but it does make the dspNexus more expensive, since this part is sort of required if you want to get the most out of the hardware.
Anyway, that's where I'm at. I've still got to get the mic out and run through another measurement session to see how it all looks, and then will probably have to tweak it a few more times until I'm happy. I'll report back once I actually get to listen to this with music.
