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dspNexus DSP Audio Processor - DANVILLE SIGNAL - SHIPPING NOW?

Why don't just use ADAU1467 as DSP?
The DSP for the next version (free upgrade to our early adopters) will use a SHARC ADSP-21569. This runs at 1GHz and has a FIR coprocessor that can do 4 billion MACs per second independently from the core. The sigmaDSP ADAU1467 runs at 294 MHz and is less powerful than the current SHARC ADSP-21469 used in the dspNexus.

Al Clark
Danville Signal
 
Haven't heard music through this yet. First I need to figure out how to convince my wife that all the cables and amps needed to run an active system like this are actually a *good* thing and make the house look better.
My solution to this problem was to build a multi-channel amplifier using Tom Christiansen's Neurochrome modules and mounting them all in one chassis. I used Neutrik 8-core speaker cable, most of which I was able to run under the floor.
 

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I am one of the early adopters of the Danville dspNexus 2/8 and want to give you some impressions. Before doing that, I can tell you that what I don’t know about setting up filters and other parameters in DSP is far more than what I do know. I have been working with DSP for many years to get the best performance out of the speakers I build. In the early days I used miniDSP and Behringer equipment. I then progressed to the DEQX Express II which was a big improvement. A large part of that was the DEQXpert service DEQX has to aid in setting up the system. In the US, their expert is Larry Owens of KRC-US, who helped me when I originally had the system using my Bohlender Graebener RD75s (January and February 2001 audioXpress magazine) with dipole midbass arrays (June and July 2004 audioXpress magazine) and subs, and later twice when I built the “eggs” and then when I moved to a new house.



For my latest set of speakers, the “eggs” (September, October, and November 2018 issues of audioXpress magazine) the DEQX with its six channels was sufficient. When I added the distributed subs to the system (August 2022 issue of audioXpress magazine), I needed two additional channels. I had been waiting for the new DEQX Pre8 Advanced HD Active but had not heard a firm delivery date, so I looked elsewhere.



Richard Hollis of Hollis Audio labs had been speaking about Danville for years and when he mentioned the new dspNexus, I signed up for the EA program. Like DEQX, an advantage of the Nexus is that you have an expert to help you with the setup and trouble shooting. Rich took some of the basic settings from my Express II and gave me a file to load on the Nexus.



As with any early unit, there have been a few annoying things, many of which are because all my equipment is located on a rack in my basement under the dedicated audio room remote from my listening position. However, to make a long story shorter, the audio performance and adjustment capabilities of the Nexus are superb. One of the things that makes a sound system sound more real is what I call texture. With massed voices you can easily hear the differences in the individual voices. With brass instruments, the subtle harmonics of the different instruments clearly makes them sound like a group of instruments and not just a mass of sound. Likewise, strings separate and hall sounds are preserved to give the sound stage depth.



Many people equate a forward frequency presentation with detail, but the real test is to have the detail while not sounding etched. The Nexus is doing that better than anything else I have used. Of course, with great recordings the sound is totally enveloping. But the real test is with the 90% of music that I like that is average in recording quality. For example, my wife and I recently went to the local theater that was playing the live concert film of the Talking Heads “Stop Making Sense”. Before going I put that recording of the performance on my system through the Nexus and was listening at average levels measured C weighted slow with My SPL meter of around 97 dB with peaks over 110 dB, and although quite loud, it was never harsh. I don’t often listen at that level for obvious hearing protection reasons, but it was a good test. Likewise, when listening to Mahler’s 2nd, the detail in the very quiet passages is there followed by clean thunderous peaks as Mahler is wont to do.



It is still early in my use of the Nexus but I will say that I am listening more to all different types of music than I ever have. And my speakers in my acoustically treated room driven by four amplifiers totaling over 6KW are very revealing of any acoustic problems. So far, so good.
 
SpeakerTom,
Thanks for the information. If you have time can you talk about using Audio Weaver and it’s useability? Also, do you have the DSP Nexus between preamp and amps or are you using the Nexus as a “preamp” using digital in from streamer, (if you are using a digital front end).
Thanks, Ted
 
I only use a very small subset of Audio Weaver. Most of the original work was done by Rich from Hollis Audio Labs. He has been working with it for years. Mostly what I have done was to work with the filters for parametric EQ and also drive levels for each amp. As I mentioned, even with the DEQX I only used a small number of features. Once either platform was originally set up, the major part of the work was in place.

I do have an analog front end with a vinyl playback system and a custom phono stage and preamp that I built. However, the vast majority of my playback is digital, either from files on my NAS or streaming from Qobuz. I have a Roon Nucleus + that gets either my local files or streamed music. It feeds an i7 intel desktop running Windows 10 hard wired to my ethernet network that acts both as a Roon bridge and also runs Audio Weaver to make the changes that are then fed to the Nexus via a USB connection. The Nexus can take both control signals and music data over the same USB connection. You can either make changes on the fly with AW or once you have a setup you like, you can flash that to the Nexus.
 
I am Rich Hollis from Hollis Audio Labs. I designed the DSP crossover for Speakertom's eggs. Installed the EA dspNexus 2x8 in his system and it works extremely well in his system.

If anyone is interested, an EA dspNexus 2x8 system will be at CAF2023 in Rockville, MD Nov 10-12. The system will drive a pair of Magnepan MG10/QR's with redesigned DSP crossovers with two 4x8in OB servo sub towers. Will be using GaN based Class D amps from Orchard Audio for the planar speakers for a fully active system.
 
I will be joining Rich at CAF, I am the lead designer of the dspNexus. You will be able to see the crossover than Rich designed using Audio Weaver and listen to the results.
Thanks.

Al Clark
Danville Signal
 
@alc @HAL Very interested in hearing about possibilities of this with my IRS Beta. I would need some help and ears which are better than mine.....
 
@alc @HAL Very interested in hearing about possibilities of this with my IRS Beta. I would need some help and ears which are better than mine.....
After the dspNexus purchase, the speaker's existing crossover will be measured at the speaker terminals to see what the original design intended. A CLIO Pocket is sent to make the measurements. This will give the data to make a preliminary DSP version of the crossover. This is one way to start the process. Another is acoustic measurements of the drivers can be made. I guide the customer through the measurements on their speakers and then send the file needed to run the new crossover design on the dspNexus.

This will also show how well the original crossover design with the component tolerances compare between the two channels. This can be corrected in the new design to improve the imaging. Also any time delays required to time align the drivers is added.
 
It was my first experience with this type of system at CAF. The engineering behind the nexus is very solid.

Rich (@HAL) is also very knowledgeable and his support is included in the price of the unit.

I highly recommend it. Especially the new version that is soon coming out with the AK4499 DAC chips.
 
Any new DSP Nexus observations from the few users on this thread? Is there new things coming to the unit soon? Seems like a few people have new changes in the fall, winter. Very interested in purchasing, still nervous about AudioWeaver as I have zero IT education.
Thanks, Ted
 
Any new DSP Nexus observations from the few users on this thread? Is there new things coming to the unit soon? Seems like a few people have new changes in the fall, winter. Very interested in purchasing, still nervous about AudioWeaver as I have zero IT education.
Thanks, Ted
Vds, there are tutorials on using Audio Weaver on DSP Concepts and Danville Signal websites. You would be working with me, if you purchase the unit from my company to implement the DSP processing needed for your system.

I will let others speak to the sound quality of the system. Just to state it here, I am also a dspNexus 2x8 user in my audio system since I was involved with its development. This is the system I wanted to use and hear in my system.

With both the Danville Signal ADSP21569 dspBlok upgrade coming for extended processing capabilities included with the system purchase and the AKM AK4499EX DAC upgrade coming for purchase, I look to be using this system for a long time.
 
As an early adopter, I am happy to chime in on sound quality:

When I first set up the dspNexus, I used it as a DAC only. I am hoping Amir eventually measures this unit as the sound quality, at least to my ears, beat my Topping D90 and I really like my Topping kit. Would love to see if the measurements tell the same story.

This unit has had the most significant impact on the sound of my system of any upgrade I have ever done. I am using all channels (3 way speakers with subs integrated). Being able to dial in the time alignment between all the drivers in relation to the listening position creates a very detailed sound stage where you know where every instrument is.

Toss in the room correction via PEQs and you can dial in your system to as flat of a response as you want. A note on the PEQs: With most dsp units out there you are limited to 10 PEQs per channel. With the dspNexus you can add as many as the processor can handle. Of course this can be a dangerous thing, but it can be quite useful in a square room like mine.

The improvement also gave me the nudge to add a decent turntable to my setup. The dspNexus has an analog input so you can feed any analogue signal through the system and benefit from the room corrections. You can set that input to normal pre-amp or RIAA (47k load) for an MM cartridge.

I also want to add that I am using 3 different amps of varying quality: a topping PA5 for the woofers, an old school Rotel (not their best) for the mids, and a McIntosh 2270 for the tweeters . The connection to the drivers is via speaker cables and a pretty janky coupler (of the home depot wall variety). Despite that I am getting phenomenal sound quality. I will be replacing all the amplifiers with a pair of 3 channel purifi amps connecting them directly to the drivers. I can only imagine what that will do in terms of further improving the sound quality.

my $.02
 
I'm also an "early adaptor" of the new Danville DSP Nexus 2x8. I was fortunate enough to have Rich Hollis actually deliver my unit in person, and spend some time with me showing it off.

I've been using DSP for speaker crossover, speaker sound correction and room eq for some years now, and the capabilities and flexibility of this new unit is a real breakthrough at any price and a deal at $3k. I started out with the DEQX HDP-4 and latter with the miniDSP 2X4 HD. Both these devices offer a wide range of IIR and FIR filter capabilities, but at some point I ran up against their limitations both in terms of capabilities and programming flexibility. There aren't a lot of manufactures of this type of equipment and choices are very limited, until the new Nexus 2X8 came along.

The programming uses DSP Concepts' "Audio Weaver," which is a very powerful graphical DSP design program very similar to Analog Devices' "Sigma Studio." Its not hard to use but it does pretty much assume you know your way around DSP terminology. Their website offers a free trial version and a lot of very useful tutorials. But, if programming isn't your style, Danville and Rich Hollis are there to help you develop a program to fit your individual needs.

The current Nexus 2X8 is perfect for a wide range of crossover design and eq purposes. Implementations using IIR filters are pretty much without limits. But it is somewhat lacking in horsepower needed for FIR filter implementation. Many implementations won't use FIR filters so this won't matter. And, Danville is working on their upgraded DSP chip from Analog Devices that will have all the FIR filter capabilities anyone could want, which will be free to current owners.
 
Thanks JazzMan for your comments.

Since JazzMan brought it up, I want to address the signal processing engine. The dspNexus has plug in modules for the ADC, DACs and DSP. Currently, the dspNexus uses a fourth generation Analog Devices SHARC DSP. I think it is probably the most powerful one in a current production DSP crossover box. We are writing code for a new module that uses a fifth generation SHARC. This SHARC has an independent FIR accelerator that can theoretically calculate 4 billion floating point MACs per second, while the DSP core is still doing everything else at the same time.

The new DSP will increase the IIR performance about 3x and make large FIRs work really well.

IMHO, IIRs work best for basic crossover features. You can use FIRs for impulse response correction because you can approximate non causal filtering using a small amount of time delay. Regardless of your preference, the goal is to have sufficient signal processing capability to do whatever you want.

Our Early Adopters will get the new DSP module as a free upgrade. We don't want people waiting to get started.

Al Clark
Danville Signal
 
Thanks JazzMan for your comments.

Since JazzMan brought it up, I want to address the signal processing engine. The dspNexus has plug in modules for the ADC, DACs and DSP. Currently, the dspNexus uses a fourth generation Analog Devices SHARC DSP. I think it is probably the most powerful one in a current production DSP crossover box. We are writing code for a new module that uses a fifth generation SHARC. This SHARC has an independent FIR accelerator that can theoretically calculate 4 billion floating point MACs per second, while the DSP core is still doing everything else at the same time.

The new DSP will increase the IIR performance about 3x and make large FIRs work really well.

IMHO, IIRs work best for basic crossover features. You can use FIRs for impulse response correction because you can approximate non causal filtering using a small amount of time delay. Regardless of your preference, the goal is to have sufficient signal processing capability to do whatever you want.

Our Early Adopters will get the new DSP module as a free upgrade. We don't want people waiting to get started.

Al Clark
Danville Signal
Hello @alc … any plans to send a unit over to @amirm for measurements?
 
Any chance that an 8 channel input like miniDSP Flex HT(x) will be available in the near future? This would facilitate the use of Dirac Live Bass Control from PC.
 
RePhase is a very useful program for developing FIR crossovers and speaker correction filters.

The DEQX units have the advantage of including built-in proprietary software for filter design. The DSP Nexus is more like the miniDSP 2X4hd in that you have to develop your own filters using external software of your choice. In fact the miniDSP website has a lot of excellent material on how various software packages work.

There is a lot of excellent free software for developing IIR filters for crossover design and room equalization. My favorite for the latter is REW, which is perhaps the most useful audio software I've ever used! You can design room eq/house curve filters and then drop the resulting biquad filters as text files into the DSP Nexus. Similarly there are lots of programs for generating IIR biquad coefficients for crossovers that can also be directly imported into the Nexus.

Options for generating FIR crossover and speaker correction filters are much more limited. RePhase is one option that is very powerful, and also free! The downside as others have noted is its tricky to use, in large part because the developer never produced a manual, leaving it up to you to figure it out. There's an online users group which is a little disorganized and thus hard to use, but the developer has very generously responded to various questions I've had over the years.

RePhase requires you to manually flatten the frequency and phase response of the speaker, after you've taken appropriate measurements, using something like REW to generate anechoic-like measurements using a gated response. RePhase gives you what look like individual PEQ filters, and by selecting appropriate center frequencies and Q values you can flatten frequency response and phase, pretty much as accurately as you care to fiddle with it. It takes some practice I admit, but now I can achieve at least as flat responses as various automated programs I've used, using only half a dozen or less filter elements -- its not that difficult.

But the really useful feature of RePhase is that you can next generate FIR crossover filters, with pretty much any slope you care to try (I routinely use 96dB/octave slopes). As Al pointed out in a previous post, it gets harder as you work with lower frequencies because of the large number of taps required, but a low pass filter for woofers at 320Hz for example is not impractical. And, the new DSP chip should allow for pretty much any conceivable number of taps, although with an increasing delay.

And of course RePhase lets you select from various filter designs including linear phase and minimum phase.

It takes some effort to figure out how to use it, but its a very powerful program and worth the effort if you're interested in FIR filter design for audio work.
 
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