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dspNexus DSP Audio Processor - DANVILLE SIGNAL - SHIPPING NOW?

The dspNexus 2/8 has been compared to a DEQX Express II in a fully custom speaker and room system in full multiamp mode. The stock unit with AKM AK4493S DACs was a step above the sound quality of the DEQX.

The latest ADSP21569 dspBlok has been released and is in the production units being shipped from Danville Signal. If you purchase the dspNexus system from my company, then I do the DSP crossover design for the customers speaker system.

Danville Signal has released the AKM AK4499EX DAC's for production units.

The production dspNexus 2/8 with the stock AK4493S DAC's is $3000.00+s/h and the AK4499EX DAC's is $4000.00+s/h.
 
The dspNexus 2/8 has been compared to a DEQX Express II in a fully custom speaker and room system in full multiamp mode. The stock unit with AKM AK4493S DACs was a step above the sound quality of the DEQX.

The latest ADSP21569 dspBlok has been released and is in the production units being shipped from Danville Signal. If you purchase the dspNexus system from my company, then I do the DSP crossover design for the customers speaker system.

Danville Signal has released the AKM AK4499EX DAC's for production units.

The production dspNexus 2/8 with the stock AK4493S DAC's is $3000.00+s/h and the AK4499EX DAC's is $4000.00+s/h.
HAL i read on your FB that you have done some Betas. Could you please share information? you can PM me if this is too narrow of a subject
 
The stock unit with AKM AK4493S DACs was a step above the sound quality of the DEQX.

Can you provide details about how this test was conducted? I'm assuming you were using meaningful controls, and this wasn't just a casual (uncontrolled) listening session?
 
The dspNexus 2/8 has been compared to a DEQX Express II in a fully custom speaker and room system in full multiamp mode. The stock unit with AKM AK4493S DACs was a step above the sound quality of the DEQX.
Further, can you define your metrics for "sound quality"?
 
Here is an article in AudioXpress magazine August 2024 issue on APx555 measurements of the dspNexus 2/8 with AK4493S DAC's.


The AK4499EX specifications are much improved over the AK4493S.

With the new AK4499EX reconstruction filter mode has improved frequency response over the earlier filter and improves transient response.

The rest is listening to both DAC's in exactly the same units to compare sound qiality. Yes, subjective and objective observations.
 
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Here is an article in AudioXpress magazine August 2024 issue on APx555 measurements of the dspNexus 2/8 with AK4493S DAC's.


The AK4499EX specifications are much improved over the AK4499EX.

With the new AK4499EX reconstruction filter mode has improved frequency response over the earlier filter and improves transient response.

The rest is listening to both DAC's in exactly the same units to compare sound qiality. Yes, subjective and objective observations.

Your link doesn't function.

I am not following this statements: "The AK4499EX specifications are much improved over the AK4499EX."

So your statement about sound quality is a subjective impression.

Thanks.
 
The link is to the Voice Coil August Issue. You have to have a login to view it.

DAC model edited for correction.

It measured well in the article and subjective comparison impressions.
 
I received my dspNexus 2/8 earlier this week. I so far have only managed to install all the needed software and setup all the wiring. I need to view the tutorial videos still, before I play with Audio Weaver. HAL sent me the design for my JBL M2s, but I will need to measure to confirm level matching and such.

Look forward to playing around with it when I get the time, which is a bit scarce these days.

I will add though that I read the manual almost completely and I am very pleased to say it is quite informative and well written. I think Danville Audio should allow potential customers to view the manual online as I think it would potentially answer a lot of questions one might have. It was clearly written by someone engineering minded, but in simple language. Refreshing. I might go as far as to say it could be seen as good marketing material as you can see how much thought and care went into the design using the best parts available.
 
Has anyone had problems using DIRAC Live, their main speakers and at least two subwoofers where the ~ 4v outputs of the DSPNexus was not sufficient to overcome gain loss caused by DIRAC, or other room correction software?
 
Can you provide details about how this test was conducted? I'm assuming you were using meaningful controls, and this wasn't just a casual (uncontrolled) listening session?

The link is to the Voice Coil August Issue. You have to have a login to view it.

DAC model edited for correction.

It measured well in the article and subjective comparison impressions.

Unless I'm wrong, along with being a DSPNexus system integrator HAL is also a Danville reseller. What's stopping him from submitting the unit for a full review by ASR, as have numerous other DAC brands?

That would likely be the best way to validate the performance of this product, especially in light of important questions raised about designer Al Clark's Axpona test setup during passive vs. DSPNexus active crossover shootouts, but which remain unanswered, at least publicly; post #60.
 
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Can someone please explain the advantage of this unit over say, a MiniDSP? I don't see any reason this unit should be recommended over a MiniDSP. It requires AudioWeaver, which needs an annual subscription and it's notoriously difficult to use. With MiniDSP, you can use REW if you don't feel like paying for a Dirac license. Also, as far as I can tell, it uses the same SHARC DSP chips which means that the number of available biquads and FIR taps would be roughly the same. So ... I don't get it.
 
You get the AW license renewed each year as part of the purchase. You get support in your design from them as part of the cost as well. Richard has helped me a lot and did the JBL M2 replication for me. It supposedly uses a next gen Sharp DSP chip than the MiniDSP, so it is meant to have more biquads and FIR taps. It can also use REW, which I have used.

It uses nicer DAC chips, whether that makes a difference is up quantifiable. It has a volume control so it works like a "preamp" directly driving your amps.

Value is in the eye of the beholder.
 
. With MiniDSP, you can use REW if you don't feel like paying for a Dirac license. Also, as far as I can tell, it uses the same SHARC DSP chips which means that the number of available biquads and FIR taps would be roughly the same. So ... I don't get it.
I currently have no hands-on experience with DSP crossovers, or the devices using multiple DAC channels. But while there's certainly a number of design factors which must be optimized to produce a high performance DAC, the design of the DAC chips are also very important. And during talks about what is known or suspected about the design of the DSPNexus a leading DAC expert https://www.diyaudio.com/community/members/markw4.373860/ did affirm that those upgraded AKM DAC chips in the latest DSPNexus are substantially better than the ESS chips in what is otherwise arguably the best sounding and measured MCH DAC. https://www.merging.com/products/interfaces/hapi

That said, I find the failure of designer Al Clark to reply to these questions to be very troubling.
 
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The DAC's are not important. All that matters is that the DAC's are not horrible, and MiniDSP does not have horrible DAC's. Although I have personally not seen any DAC measurements of the Nexus, I would be pretty confident that it's not going to be horrible either.

What matters is the DSP capability, since that is why people buy this unit. The DSPNexus website does not say how many IIR biquads are available, or how many FIR taps ... so we don't know what it is capable of. I am sure that info is buried in a manual somewhere.
 
miniDSP doesn't have a unit with 8 channels and FIR capability. Nor does the miniDSP Eight have analogue inputs or XLR. Leaving analogue inputs out is seriously stupid and was done so they could fit it in the same chassis as the 4 channel version.

The Nexus uses a powerful processor, so it should have very high FIR capability. It also much more versail in what it can do. But I agree that Audooweaver isn't a software for the end user, and that's why I doubt the product will ever reach high sales unless Danville can make a proper front end software.

As for the DAC chips, they are better in the Nexus and many audiophiles want that despite it's unlikely it will make an audible difference.
 
Do you know how many FIR taps are available? If it's something like 4096 taps at 96kHz, it's useless so they may as well not have any. All the heavy lifting would be done by the biquads.

That's my beef with all these hardware processors - you buy them for their DSP capability, but they don't tell you what those capabilities are.
 
That's my beef with all these hardware processors - you buy them for their DSP capability, but they don't tell you what those capabilities are.
It's pretty poor but you can get a rough idea

I think it was said it uses https://www.analog.com/en/products/adsp-21569.html at 1GHz core clock, it does 4 mac operations per cycle so 1GHz *4 / 48KHz = ~83333 taps assuming perfect efficiency and doing nothing else but fir. For 8 channels that's in the region of 10k taps per channel so better than most but still not a massive number. If they use the same chip for the lower channel count variants then multiply those numbers by 2 or 4 per channel depending on the channel count.

Arguably it's sufficient for a woofer in a 4 way but not the sub whereas competing models wouldn't even manage the woofer.
 
Do you know how many FIR taps are available? If it's something like 4096 taps at 96kHz, it's useless so they may as well not have any. All the heavy lifting would be done by the biquads.

That's my beef with all these hardware processors - you buy them for their DSP capability, but they don't tell you what those capabilities are.
Considering it's using ADSP-21569, I'm assuming not more than 4096 taps per channel. Or perhaps more per channel if all 8 channels aren't utlized, or by breaking the computation into smaller blocks.

4096 taps is the same the miniDSP Flex (4-channel unit) has per channel. Useless? I highly disagee. First of all, IIR can be used to achieve linear phase. But obviously using FIR is a lot easier. But I don't believe the more the merrier here. I have listened to a large number of speakers with a high number of FIR taps, including around 8 setups with Audiolense. None of the Audiolense setups sounded anything near to a great setup with only IIR. They were all really poor sounding actually.

Getting the basics done first is really the most important factor. (Read: Speaker design and acoustics). High amount of FIR taps can't save something that's partially broken in the first place. Many of the setups that uses a lot of taps typically uses correction in a wrong matter also IMO, and the result is always very unatural sound with phase distortion. So I'll rather take a good IIR setup over a poor one with FIR anyday.
And 4096 per channel works really well.
 
Considering it's using ADSP-21569, I'm assuming not more than 4096 taps per channel. Or perhaps more per channel if all 8 channels aren't utlized, or by breaking the computation into smaller blocks.

4096 taps is the same the miniDSP Flex (4-channel unit) has per channel. Useless? I highly disagee. First of all, IIR can be used to achieve linear phase.

I am sorry, but this is incorrect. You can't have linear phase with IIR, because linear phase requires symmetrical mirror image zeros outside the Z-plane unit circle. That is not possible with IIR, all the zeros have to be within the unit circle. It is possible to have linear phase regions of the frequency response, but all XO's and all PEQ's are minimum phase.

4096 taps is useless, unless you are able to allocate those taps to frequency bands, e.g. with a multirate polyphase system to increase the apparent resolution. At 96kHz, 4096 taps gives you a frequency resolution of 23Hz. I don't know what you can do with bins that large.

Listening to a speaker with a bad FIR filter and not liking it is not evidence of anything. There are good and bad implementations and it depends on user skill. I have heard dozens of very bad speakers with passive XO's, but I wouldn't be so quick to dismiss all passive XO's as bad. I think that any speaker with a passive XO hasn't reached its full potential, but that does not mean that it's not good as it is.
 
I am sorry, but this is incorrect. You can't have linear phase with IIR, because linear phase requires symmetrical mirror image zeros outside the Z-plane unit circle. That is not possible with IIR, all the zeros have to be within the unit circle. It is possible to have linear phase regions of the frequency response, but all XO's and all PEQ's are minimum phase.

4096 taps is useless, unless you are able to allocate those taps to frequency bands, e.g. with a multirate polyphase system to increase the apparent resolution. At 96kHz, 4096 taps gives you a frequency resolution of 23Hz. I don't know what you can do with bins that large.

Listening to a speaker with a bad FIR filter and not liking it is not evidence of anything. There are good and bad implementations and it depends on user skill. I have heard dozens of very bad speakers with passive XO's, but I wouldn't be so quick to dismiss all passive XO's as bad. I think that any speaker with a passive XO hasn't reached its full potential, but that does not mean that it's not good as it is.

IIR can be used to achieve linear phase crossovers, but it isn't trivial.

One way is using APFs -> https://www.diyaudio.com/community/...hase-linearization-by-fir.393435/post-7835131, but as you can see coming up with an automated way to calculate those APFs isn't easy.

If you have a processor that can implement subtractive filters you can also do it with only IIR and delay -> https://www.audiosciencereview.com/...ding-subwoofer-integration.58232/post-2131400, I bet the Danville can do this but miniDSP cannot.

Michael
 
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