DEQX Premate 8 digital active crossover / DSP

[mdsimon2]

I've already covered my thoughts on minDSP vs DEQX vs CamillaDSP/software DSP elsewhere in this thread, but just wanted to point out that CamillaDSP can easily handle inputs besides USB.

However - it is DIY, has no built-in DAC's, and only USB input.

I think this is one of CamillaDSP's great strengths, you can easily use different capture and playback devices AND implement adaptive asynchronously resampling to avoid buffer under/over runs even if those devices are not clock sync'd.

Michael

 

[Sined]

You are essentially depending on the algorithm to make decisions for you. What if you don't like those decisions? You would hope there is some way to over-ride it and make the software do what you want. To me this is the biggest unknown of the DEQX.

Yep !
This is why I choked to jump into the new DEXQ pre-8 acquisition and testing. I'm afraid that, to reach more non-technical users and to lower after sales support costs, the software will be developed to be more user friendly and possibly do a lot of calibration automatically behind the scene and for which we have no control for fine tuning. I prefer to wait and see a user's manual to see how the engine will work.

 

[sarieri]

I've already covered my thoughts on minDSP vs DEQX vs CamillaDSP/software DSP elsewhere in this thread, but just wanted to point out that CamillaDSP can easily handle inputs besides USB.



I think this is one of CamillaDSP's great strengths, you can easily use different capture and playback devices AND implement adaptive asynchronously resampling to avoid buffer under/over runs even if those devices are not clock sync'd.

Michael

I think Camilladsp is pretty much an end game DSP except that it cannot decode various multichannel audio format. But that’s the problem for most DSPs unless you go to some AVR route.

 

[Keith_W]

I think Camilladsp is pretty much an end game DSP except that it cannot decode various multichannel audio format. But that’s the problem for most DSPs unless you go to some AVR route.

I would be interested in Camilla, except that the install process is horrendously complex on a Windows PC. It required installation of some library, and then it required a few commands to be typed in via a command line before it would launch. I couldn't get it working, a friend who is a computer programmer (and who had managed to install Camilla) had to come over and even he had trouble. It took him hours. After he left, I couldn't figure out how to launch it. I never got past the install process, so I have a number of questions about whether it is worth trying again.

- Can it host VST's?
- Can you switch between filters via a web app? I use Acourate Convolver and this is my favourite feature.
- What is the latency between filter changes? Is it a zero latency filter like Hang Loose Convolver?
- How easy is it to load filters? Is a command line required?

If it ticks all those boxes, then I will consider trying to install Camilla again.

 

[3ll3d00d]

- Can it host VST's?

no - see https://github.com/HEnquist/camilladsp/issues/74

Can you switch between filters via a web app?

there is a web ui that lets you load different configs, as far as I recall that means restarting the audio pipeline so it's never going to be zero latency.

as an aside, on windows the main problem, for me at least, with camilla is that it doesn't support asio directly.

 

[mdsimon2]

I would be interested in Camilla, except that the install process is horrendously complex on a Windows PC. It required installation of some library, and then it required a few commands to be typed in via a command line before it would launch. I couldn't get it working, a friend who is a computer programmer (and who had managed to install Camilla) had to come over and even he had trouble. It took him hours. After he left, I couldn't figure out how to launch it. I never got past the install process, so I have a number of questions about whether it is worth trying again.

- Can it host VST's?
- Can you switch between filters via a web app? I use Acourate Convolver and this is my favourite feature.
- What is the latency between filter changes? Is it a zero latency filter like Hang Loose Convolver?
- How easy is it to load filters? Is a command line required?

If it ticks all those boxes, then I will consider trying to install Camilla again.

I understand CamillaDSP is not for everyone, I just don't appreciate spreading blatantly false information like saying it only supports USB inputs.

Just my 2 cents but if I wanted to use CamillaDSP and I was starting from a Windows PC, I'd get a raspberry pi and use it in USB gadget mode. In such a setup the raspberry pi running camillaDSP acts as a USB I/O device. To a Windows PC the raspberry pi looks like any other USB audio device, and you can set it up to use any number of input / output channels. Installation on Linux is easy if you can follow instructions and copy / paste.

There is a web app, after installation you use the web app to control everything in a configuration, no need to touch the command line after installation.

You can easily switch filters in the web app. For short filters the switch is instantaneous with no perceptible delay. If you are implementing linear phase FIRs that are several seconds long there is a slight pause between switching filters (around 0.5 sec).

Michael

 

[Ibanez]

Danville Nexus, complete DSP unit. 48V phantom powered mic, USB in, 2 XLR inputs, 8 XLR outputs, DSP Concepts' Audio Weaver, that one in-box solution i would have considered if i didn't have to import it to Sweden, that will be $3000 x 1.25 (vat deposit) = 3,794. Just too much. PC with Acourate and my used Antelope ossc is cheaper, and more channels to play with also. But i really like the idea of their Audio Weaver, you can time alignment the drivers in real-time in that program, which not acourate, audiolense or dephoinca will do. Heck, even my old car stereo Pioneer Deh-P88RSII can do that and it's from what, 2008, why cant modern programmed PC based DSP:s do that? so much cpu power we have in todays pc's that should not be to hard to implement, if an old car stereo unit could do it from 2008, and even older units than that could do it in real-time, why not.
I wish i could buy an Audio Weaver license for private use, but as i understand that is not possible due to it is hardware specific.

 

[Tranquility Bass]

Danville Nexus, complete DSP unit. 48V phantom powered mic, USB in, 2 XLR inputs, 8 XLR outputs, DSP Concepts' Audio Weaver, that one in-box solution i would have considered if i didn't have to import it to Sweden, that will be $3000 x 1.25 (vat deposit) = 3,794. Just too much. PC with Acourate and my used Antelope ossc is cheaper, and more channels to play with also. But i really like the idea of their Audio Weaver, you can time alignment the drivers in real-time in that program, which not acourate, audiolense or dephoinca will do. Heck, even my old car stereo Pioneer Deh-P88RSII can do that and it's from what, 2008, why cant modern programmed PC based DSP:s do that? so much cpu power we have in todays pc's that should not be to hard to implement, if an old car stereo unit could do it from 2008, and even older units than that could do it in real-time, why not.
I wish i could buy an Audio Weaver license for private use, but as i understand that is not possible due to it is hardware specific.

You can run Audioweaver on a PC and I believe you can get a free trial to evaluate it

 

[Keith_W]

Danville Nexus, complete DSP unit. 48V phantom powered mic, USB in, 2 XLR inputs, 8 XLR outputs, DSP Concepts' Audio Weaver, that one in-box solution i would have considered if i didn't have to import it to Sweden, that will be $3000 x 1.25 (vat deposit) = 3,794. Just too much. PC with Acourate and my used Antelope ossc is cheaper, and more channels to play with also. But i really like the idea of their Audio Weaver, you can time alignment the drivers in real-time in that program, which not acourate, audiolense or dephoinca will do. Heck, even my old car stereo Pioneer Deh-P88RSII can do that and it's from what, 2008, why cant modern programmed PC based DSP:s do that? so much cpu power we have in todays pc's that should not be to hard to implement, if an old car stereo unit could do it from 2008, and even older units than that could do it in real-time, why not.

"Real-time" adjustment of delays is a function of the convolver, not the filter design software (which is what Acourate and Audiolense are designed to do). Nearly all convolvers I can think of have the ability to adjust delays, though not all of them can do it real-time. It may involve stopping the music and reloading the filter.

I am curious how "real-time" delay adjustment in Audio Weaver works. Would you be able to describe the procedure?

 

[Ibanez]

You just turn an knob in software i suppose, not very advanced, and it changes on the-fly ,like on my P88RSII (P880 in US) car stereo, it changes on the fly, no need to restart music. Should not be nedded with an PC, with all the cpu it has today. That worked 2008 on that machine. Why do todays convulsions need restarting, when i shouldn't really be nedded?
And i ask to learn.

Dephonica doesn't need to restart niether, there you just press return when given numbers in meters/cm/mm are written, and changes, but in DspNexus there it is an knob as i understand, that you just turn in software and it changes directly, which is cool, and it's much easier to calibrate the alignment in that way, especially if you use whitenoise or pinknoise.
And AVRs can also do it in direct on the-fly change.

 

[Ibanez]

You can run Audioweaver on a PC and I believe you can get a free trial to evaluate it

But can you buy it as end customer?
I believe it's only for oem, and you only trial for private use, but i can have mistaken me on that one.
I will check again

 

[Ibanez]

Hello again, was on dsåconcpts homepage, and they require company tp even register on their page.
So i believe it's aimed only towards oem/odm manurfacturers.
Thats a little sad, but it's done on units like the dspnexus, not for actual pc use as we want to use it, like Acourate, Audiolense or Dephonica.
It's a pity that dephonica doesn't have more than eight output channles, i would have like only two or four more, 4-way with two or more subs, all time aligned.

 

[Tranquility Bass]

Hello again, was on dsåconcpts homepage, and they require company tp even register on their page.
So i believe it's aimed only towards oem/odm manurfacturers.
Thats a little sad, but it's done on units like the dspnexus, not for actual pc use as we want to use it, like Acourate, Audiolense or Dephonica.
It's a pity that dephonica doesn't have more than eight output channles, i would have like only two or four more, 4-way with two or more subs, all time aligned.

The other issue is that when I bundled Audioweaver with the preamp I was selling, people had to pay a subscription of $100 US a year to keep using Audioweaver whilst I was absorbing the initial upfront cost but I guess 12 months is probably enough time to nut out a crossover design. However for most people that this type of product appeals to tend to continuously like to tinker with their systems.

cheers

 

[Ibanez]

The other issue is that when I bundled Audioweaver with the preamp I was selling, people had to pay a subscription of $100 US a year to keep using Audioweaver whilst I was absorbing the initial upfront cost but I guess 12 months is probably enough time to nut out a crossover design. However for most people that this type of product appeals to tend to continuously like to tinker with their systems.

cheers

Okey, i thought that DspNexus was not an subscription model / year, it does not say that om their homepage. And isn't Audioweaver on the likes on DspNexus also an convolver for Fir/iir filters and taps and things like that, not just the crossover filters?
Or do they have like several licenceing models depending on usercases. Because, else the Dspnexus is way much more in price than the $3000 advertised if it's an yearly subscription of $100/year, what in 10 years, the that unit would be more hefty to buy than likes Merging Hapi/Horus or other Ravenna/Dante dacs in conjunction with Acourate that only will be an one-time fee, even if pretty hefty. Hapi is pretty expensive even without the DA8P card, but if DspNexus turns into an subscription DAC/DSP it will get way out my thoughts to buy.
Cool that you actually had tried Audio Weaver, it's nice to hear from an actual licenced user of it.
Thanks for sharing it.

 

[Tranquility Bass]

Okey, i thought that DspNexus was not an subscription fee/yea, it does not say that om their homepage. And isn't Audioweaver on the likes on DspNexus also an convolver for Fir/iir filters and taps and things like that, not just the crossover filters?
Or do they have like several licenceing models. Because, else the Dspnexus is way much more in price than the $3000 advertised if it's an yearly subscription of $100/year, what in 10 years, the that unit would be more hefty to buy than likes Merging Hapi/Horus or other Ravenna/Dante dacs in conjunction with Acourate that only will be an onegtime fee, even if pretty hefty. Hapi is pretty expensive even without the DA8P card, but if DspNexus turns into an subscription DAC/DSP it will get way out my thoughts to buy.
Cool that you actually had tried Audio Weaver, it's nice to hear from an actual licenced user of it.
Thanks for sharing it.

All I can say is that my customers were all paying dspconcepts to renew their licenses so I told them to only renew when they needed to. Normally, dspconcepts charges a small amount for the runtime libraries used on an end product where only the OEM has access to the Audioweaver design software to design the product so my use case was different where the customer of the preamp required Audioweaver to customize it for their own use. Still when you consider how much time it saves you from having to learn how to code a DSP I don't think many can really complain about the price. I can't say that this agreement is the same for other vendors in the same situation as myself but I would be surprised if it wasn't.

cheers

 

[Ibanez]

And if we compare to Okto Dac8 Pro, €1611, custom volt €1111, remote. Amd i don't remember if those prices is with VAT,
Dedicated PC €1000, Acourate Toolbox and Convolver €500, = €3722 without Windows 11 and Jriver licenses.

Then DspNexus, if there is no subscription model on that unit, it's actually pretty cheap for what is does, and you actuallly don't need any dedicated PC to use it, except when configuration on filters, measurements, and that kind of stuff. But after those steps is done, it is an standalone unit, as i have understood it, you can actually just connect your phone with usb cable and use like USB Audio Player Pro to play lossless with digital active crossovers. Pretty cool, but too expensive with import fees added (25 tax and freight costs).

But thanks again for sharing your actual use of the program, a real user that is talking about it, thanks again.

 

[Keith_W]

And if we compare to Okto Dac8 Pro, €1611, custom volt €1111, remote. Amd i don't remember if those prices is with VAT,
Dedicated PC €1000, Acourate Toolbox and Convolver €500, = €3722 without Windows 11 and Jriver licenses.

Don't use an Okto DAC8 Pro, you can't take measurements with it. It has no microphone input. You need a device like a Motu Ultralite Mk.5, RME Fireface UC/UCX/Babyface, Merging Anubis/Hapi/Horus, etc. Some cost less than the Okto (the Motu and the Anubis), some cost more (Hapi and Horus). The RME is roughly the same price whilst offering superior functionality.

Don't forget to factor in the cost of a calibrated microphone, mic tripod, cables, and additional amplifiers. Also, none of these pro audio interfaces have standard XLR/RCA outputs, so factor in the cost of D-sub breakout cables or 1/4" TRS cables. You will be very unpleasantly surprised when you find out how much a stupid D-sub breakout cable from Merging costs, you could buy a Topping DAC for that kind of money.

One big plus of the new DEQX Premate is that it is the only hardware convolver on the market I am aware of that is capable of lin phase FIR filters. None of the others have this capability. Everything else is IIR or mixed phase (FIR on top, IIR on bottom). This is doubly disappointing when you realize that a Raspberry Pi handily beats those SHARC DSP units when it comes to CPU power, and Raspberry Pi's aren't expensive. Even the new ADSP-21569 chips don't have enough grunt to run FIR filters. You can run lin phase FIR filters on a Pi with CamillaDSP, then you can save yourself €1000 on a dedicated PC.

To be fair, FIR and IIR filters have pros and cons. IIR filters can not be lin phase, they are min phase only. They require less computing resources, and generally have lower latency. FIR filters can be either lin phase or min phase, but have the potential to suffer from ringing. The point is, if you have CPU power to spare, you can choose. If you bought a unit which is CPU limited from the beginning, you have no choice. If you are happy to stick with IIR min phase filters, get a MiniDSP. Everything else on the market offers only incrementally more computing power than a MiniDSP whilst requiring horrendously unfriendly software which requires a subscription to keep working.

 

[Tranquility Bass]

One big plus of the new DEQX Premate is that it is the only hardware convolver on the market I am aware of that is capable of lin phase FIR filters. None of the others have this capability. Everything else is IIR or mixed phase (FIR on top, IIR on bottom). This is doubly disappointing when you realize that a Raspberry Pi handily beats those SHARC DSP units when it comes to CPU power, and Raspberry Pi's aren't expensive. Even the new ADSP-21569 chips don't have enough grunt to run FIR filters. You can run lin phase FIR filters on a Pi with CamillaDSP, then you can save yourself €1000 on a dedicated PC.

What a load of nonsense ! If you want a dedicated DSP then you use a proper DSP and not a phone chip !! Do you have any evidence that the ADSP-21569 "doesn't have enough grunt to run FIR filters" or did you read that off the back of a corn flakes packet ?? Please provide it then. Considering that the previous DEQX models used a 1st gen SHARC chip from the 90's and you having claimed to have owned this product at some stage it's a bit rich of you now bagging a 5th gen part just because DEQX chose not to use it ! And if the rumours are true that minidsp are designing in a 5th gen part well it's pretty much game over for the general purpose customizable DSP box. Well and truly commoditized ! Also here are three good reasons NOT to use linear phase filters and/or group delay correction when building a loudspeaker !!

https://www.grimmaudio.com/wordpress/wp-content/uploads/speakers.pdf

And the take-home message from this document is:-

QuoteDSP loudspeaker crossovers done right

From the above we've learned that:-

• Heavy-handed correction exacerbates acoustical problems
• Sharp, linear-phase filters cause pre-ringing
• Targeting an exact linear-phase sum can cause pre-echos.

In short, brute-force correction sounds grainy and smudgy. When you hear cymbals go "splash" instead of
"crash", it's naive DSP at work. So:-

• Do not shave off the hair, a nasty stubble will grow back.
• Do not correct beyond the very beginning of the impulse response.
• The gentler you correct, the wider the angle over which the correction still improves things.
• Target a minimum phase sum.

For the time being I would strongly recommend designing the correction manually. This rules out FIR as the
main workhorse. For each bump or dip one corrects, one should know exactly where it comes from, and make
sure that it isn't better corrected for acoustically. Unfortunately, designing DSP filters does not relieve one from having to know one's acoustics.

Also:-

https://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf

QuoteAlso shown in Figure 8 is the impulse responses for a complementary high-pass brick-wall FIR. If we construct a crossover filter bank with such a complementary pair of filters, the Gibbs ripples are also complementary (since we will expect the filters to sum to a flat response with linear phase, producing a perfect impulse). The summed result will thus be free of any Gibbs ripple, so what's the problem? The problem is off-axis. The complementary ripples will only cancel if the delay suffered by the signal from each driver is identical. Off-axis, where the path lengths differ, the ripples will not cancel, leading to the possibility that Gibbs ripple might become audible (just like a high-Q ringing filter).

Such summing errors will be more pronounced at higher crossover frequencies because the ripples are more closely spaced. Lower frequency
crossovers will have wider ripples which will more easily cancel in the presence of off-axis induced delays.

It is evident that steeper cut-off slopes give rise to Gibbs ripples of greater duration. It makes sense therefore to restrict the cut-off slope to be no more than is necessary for the application.

And from the Linkwitz Lab website pretty much the same cautionary tale

https://www.linkwitzlab.com/frontiers.htm

I - Digital crossovers

Some people think that digital crossovers will replace analog ones, because digital filters can be designed with desirable characteristics that are impossible to realize with analog circuitry. In particular, lowpass and highpass filters with extremely steep slopes and linear phase shift are possible. Steep slopes reduce the overlap region between drivers. Linear phase shift eliminates waveform distortion and merely causes a delay of the signal. Such characteristics can be obtained from the digital equivalent of tapped delay line filters, which have a finite impulse response (FIR) duration that depends upon the number of taps used. Digital FIR filters can have almost any desired frequency response, if the number of weighted taps is made sufficiently high. [1]

The linear phase shift comes at a price. The impulse response rings. The more so, the steeper the filter slopes. Both lowpass and highpass sections of the crossover ring, but when the outputs are combined, as for a crossover, then the two impulse responses add to a non-ringing, delayed pulse.

All would be fine, if we listened only in anechoic spaces or to speakers with coincident drivers. In reality we use speakers in rooms with reflections and reverberation and the the drivers are separated from each other due to their sizes. As a consequence the off-axis response of the speaker matters and contributes to what we hear. With the drivers non-coincident, the lowpass and highpass outputs are delayed different amounts at points off-axis, and the ringing is no longer canceled in the addition. In the best case the drivers might be coaxial, but this has another set of problems. Very steep crossovers can also cause a very abrupt change in the polar pattern of the speaker, when transitioning from a large diameter driver to a small one. Under reverberant conditions and/or listening off-axis this may have audible consequences.

 

[Keith_W]

The new DSP Nexus uses the ADSP-21569 chip, and it can not do FIR filters. If you are aware of any FIR capable hardware convolver that uses the ADSP-21569 I would be quite interested. I have been looking for a FIR hardware convolver for quite some time.

I knew that you would pull out that Putzeys paper, it seems to be a favourite of yours. Ringing is risk that you run with FIR filters, you can certainly produce filters that don't ring, or ringing that is so low that it is inaudible. Compare this to min phase which rotates phase. It effectively takes your signal and smears it across time.

You seem to have truncated the Linkwitz quote. For the benefit of everyone reading, this is the full quote:

The linear phase shift comes at a price. The impulse response rings. The more so, the steeper the filter slopes. Both lowpass and highpass sections of the crossover ring, but when the outputs are combined, as for a crossover, then the two impulse responses add to a non-ringing, delayed pulse.

All would be fine, if we listened only in anechoic spaces or to speakers with coincident drivers. In reality we use speakers in rooms with reflections and reverberation and the the drivers are separated from each other due to their sizes. As a consequence the off-axis response of the speaker matters and contributes to what we hear. With the drivers non-coincident, the lowpass and highpass outputs are delayed different amounts at points off-axis, and the ringing is no longer canceled in the addition.

Linkwitz does not mention that it is possible to digitally delay drivers so that they can become coincident again. And, as Putzeys points out, the ringing is not audible.

And BTW, you seem to have quite an irrational hatred of DEQX (and also a few other things, e.g. Elektra). I wonder what they ever did to you.

 

[Tranquility Bass]

The new DSP Nexus uses the ADSP-21569 chip, and it can not do FIR filters. If you are aware of any FIR capable hardware convolver that uses the ADSP-21569 I would be quite interested. I have been looking for a FIR hardware convolver for quite some time.

I knew that you would pull out that Putzeys paper, it seems to be a favourite of yours. Ringing is risk that you run with FIR filters, you can certainly produce filters that don't ring, or ringing that is so low that it is inaudible. Compare this to min phase which rotates phase. It effectively takes your signal and smears it across time.

You seem to have truncated the Linkwitz quote. For the benefit of everyone reading, this is the full quote:



Linkwitz does not mention that it is possible to digitally delay drivers so that they can become coincident again. And, as Putzeys points out, the ringing is not audible.

And BTW, you seem to have quite an irrational hatred of DEQX (and also a few other things, e.g. Elektra). I wonder what they ever did to you.

I never truncated anything. That's what the forum software does. What you quoted from Linkwitz is exactly what I quoted if you click on the pulldown to expand it. Obviously it comes down to who are you going to believe. A qualified engineer with years of experience or a part time audio hack who knows how to press a few buttons ??

Instead of shooting the messenger why don't you step up to the plate and provide evidence to the contrary and prove that the SHARC chip can't do FIR or convolution in hardware when the block diagram for the chip clearly states it has dedicated hardware to do this. Regarding DEQX why don't you tell us what phone chip they are using otherwise the argument is pretty one sided. Any part numbers otherwise you are making baseless claims based on insufficient information or advertisers blurb ??