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Digital Crossovers Software Hardware technical questions(Speaker Management Systems)

Severian

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I would like to switch from my post-DAC/preamp DSP crossover to something in software so as to avoid the additional ADC/DAC step. The toughest nut to crack for doing this affordably seems to be multichannel volume control. Very few pro audio interfaces allow their monitor volume knobs to control more than two output channels. Multichannel audiophile DACs with volume control are very expensive. I would be fine controlling the volume via my PC, but I want some kind of knob and the ones I've seen seem to control the Windows audio mixer whereas I would be using ASIO.
 
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Trdat

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somebodyelse

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I would like to switch from my post-DAC/preamp DSP crossover to something in software so as to avoid the additional ADC/DAC step. The toughest nut to crack for doing this affordably seems to be multichannel volume control. Very few pro audio interfaces allow their monitor volume knobs to control more than two output channels. Multichannel audiophile DACs with volume control are very expensive. I would be fine controlling the volume via my PC, but I want some kind of knob and the ones I've seen seem to control the Windows audio mixer whereas I would be using ASIO.
MOTU publish the network control interface specs for their AVB products, so it would be relatively easy to make a control interface to adjust multiple output levels from a single control. This could be an app, a web page, or a hardware device depending on preference and/or skill.
 

ofrappier

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I use OSC command and OSCULATOR software on MAC OSX , OSC to MIDI : and i control multiple output levels, with an AKAI APC MINI.

I control the "trim'" of the MOTU OUTPUT, not the DIGITAL VOLUME of the internal MIXER.

:)

I will post OSC command to MIDI, soon

Regards,

Olivier F.
 

Kal Rubinson

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I would like to switch from my post-DAC/preamp DSP crossover to something in software so as to avoid the additional ADC/DAC step. The toughest nut to crack for doing this affordably seems to be multichannel volume control. Very few pro audio interfaces allow their monitor volume knobs to control more than two output channels. Multichannel audiophile DACs with volume control are very expensive. I would be fine controlling the volume via my PC, but I want some kind of knob and the ones I've seen seem to control the Windows audio mixer whereas I would be using ASIO.
1. See my latest (last) MITR for the Merging Anubis.
2. Depending on your chosen music player (I use JRiver), you can add a VC to it. I love this one:
 

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Olli

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Would Audiolense correct clock drift or timing issues with different DACS?

@Trdat, I have an RME ADI-2 Pro and a Benchmark DAC3 combined for runing a 2.4 set up as a virtual 3 way speaker (2 subs < 62 Hz, 2 subs 66 - 165 Hz, Mains < 165 Hz) system with Audiolense. 4 subs are with the RME, the mains are with the DAC 3. Here's the Feedback I got from Benchmark on the matter of clock drift:

"They [the clocks] will both be synchronized by the RME, but the DAC3 will add a small amount of delay, but this should not be enough to matter when time aligning a sub to the rest of the system. The delay added by the DAC3 is a function of the sample rate. The delay is reduced at higher sample rates. See the chart on page 65 of the DAC3 HGC manual:

https://cdn.shopify.com/s/files/1/0321/7609/files/DAC3_Series_Manual_Rev_C.pdf?9754597140500673782

At a 44.1 kHz sample rate the delay is 1.36 ms. At 192 kHz, the delay is 0.47 ms. To put this into perspective, sound travels about 1.1 foot in a millisecond. The net effect is equivalent to moving the sub about 1 foot closer to your listening position.

Here is another way of looking at the problem: At a crossover frequency of 80 Hz, the wavelength is 13.75 feet. Full cancelation between the mains and subs would occur if the sub was 13.75/2 feet from the mains (either in front or behind). It would also occur in the time delay difference was 1/80 Hz = 12.5 ms. At the worst-case 1.36 ms delay, the phase error is 19.6 degrees at 80 Hz.

However, the delay produced by the DAC3 will be offset by the delay produced by the D/A converter in the RME. I suspect that the RME produces slightly less delay than the DAC3. The net difference between the output of the DAC3 and the D/A in the RME will be much less than 1 ms and this should be insignificant in a subwoofer application."

On top of this from what I understand Audiolense will correct delays between all channels anyhow; so not a problem.

My recommendation for a very versatile and affordable MCH DAC that includes ADC and a mic pre as well (this makes life with Audiolense much easier vs using a USB Mic such as MiniDSP UMIK-1) is a MOTU Ultralite MK4.
 

Severian

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Until this thread, I was not aware of J River's WDM driver that enables its DSP stack to be applied systemwide. In conjunction with that USB volume knob controlling J River's volume, that might do the trick for me.
 
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Trdat

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On top of this from what I understand Audiolense will correct delays between all channels anyhow; so not a problem.

I am under impression that the time aligning from Audiolense is for sound coming from different drivers not from different resampling rates. After an Audiolense sweep I can't imagine the sweep will detect the difference in the continuing sampling rates from a DAC. It might initially but sampling rates will probably change consistently and inconsistently But please correct me if I am wrong?

If I have understood you correctly your saying that for sub integration a different DAC wont make that much difference which is good to now, I suppose the only way to find out is to try and see if music prevails over the other.

I to want to use one of my expensive DAC for Sub integration it has great impact and punch and keen to use it for the sub woofer while using my newly purchased multi channel DAC for the rest of my 2 way system.(Tweeter and Mid will keep the passive crossover) so very keen to know if different DACs work...

My recommendation for a very versatile and affordable MCH DAC that includes ADC and a mic pre as well (this makes life with Audiolense much easier vs using a USB Mic such as MiniDSP UMIK-1) is a MOTU Ultralite MK4

Well, I already bought the MiniDSP DAC8 so it will have to do for now, but I do have a external soundcard that I use for measuring so it will work as easy as a mic pre on the MOTU.
 

Jukka

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I must add, if using digital/active xovers, one can use lower power (=cheaper) amps, because there is no passive components to eat power, and tweeter and mid range takes a lot less power than woofers anyway.

If using reasonable priced devices and/or DIY amps, the total cost can even be lower than when using good quality (=expensive) passive xover components, especially with 3- or 4-way speakers. Some 4-way Troels speakers can cost 3k just in the xover.

With DIY, the story doesn't usually end with one pair of speakers, so reusing the xover and amps can save a bunch of money. Also as digital filters are easy to change, development can be easier, cheaper AND quicker!
 

Jukka

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As for devices?

I'm currently considering (as European) Thomann offerings, t.racks DSP 408 and FIR DSP 408. These are post-volume-control AD-DA, but is that so bad? Options are few and far between if looking for perfect solution (digital all the until da and amps, volume control for all channels at latest possible moment, display for volume, multiple digital inputs). The closest I've found so far is miniDSP 4x10 HD fed by SHD (Studio). Any other ideas?

For amps I got existing class d for woofers, but what for tweeter and mid range? Maybe some LM3886 based DIY?

-- EDIT I just noticed that the FIR DSP 408 has a digital input. We may have a problem solver on or hands. We would need a digital preamp thou..
 
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Ro808

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Jukka, I am using a similar setup like you're considering.
Digital out from an USB DAC > loudspeaker management system > 2 separate power amps > 2 way loudspeakers.
There's only one DA conversion in the signal path, which I consider mandatory with regard to sound quality.

The digital pre amp could also be a digital mixer.

Honestly, I do have my doubts about the engineering of the t.racks gear, as this user comment on the DSP 408 (without FIR) illustrates:

"Has everything a processor needs, just 1 big loss ...
The delay time on the outputs is in steps of 5 ms. These are far too big steps to correct your system (top's and sub's) in time. If the steps would decrease to 1ms, this is really a top machine. Please update!
In terms of sound, it is doable. A Lake obviously sounds better."
 
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dc655321

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Stumbled on a review of the T.Racks fir dsp 408.
More from a PA viewpoint but still interesting I think.


That report indicates the device offers only 4k taps for all FIR filters.
If one was leaning in that direction, this is major limitation.
 

Jukka

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Jukka, I am using a similar setup like you're considering.
Digital out from an USB DAC > loudspeaker management system > 2 separate power amps > 2 way loudspeakers.
There's only one DA conversion in the signal path, which I consider mandatory with regard to sound quality.

The digital pre amp could also be a digital mixer.

Honestly, I do have my doubts about the engineering of the t.racks gear, as this user comment on the DSP 408 (without FIR) illustrates:

"Has everything a processor needs, just 1 big loss ...
The delay time on the outputs is in steps of 5 ms. These are far too big steps to correct your system (top's and sub's) in time. If the steps would decrease to 1ms, this is really a top machine. Please update!
In terms of sound, it is doable. A Lake obviously sounds better."
I didn't see this message before. Must have slipped my eyes.

FWIW, I went for miniDSP SHD and Hypex FusionAmps, the latter having only IIR filters, but that's not bad.

t.racks processor also has a fan, which is a deal-breaker for home use. Would have needed modding. The SHD just has all features that home user needs, so I decided to invest with money this time. And the Hypex plates are also fantastic with auxiliary features, like good connectivity, signal sensing automatic standby etc.
 
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