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Pls help with 3 box design... 5 years 1 month ago #26701

  • captgnarly
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Hi all, here is what I think I want to do...subject to advice of course :)

Objective is a 4-way active stereo system, (sub-MF-HF-VHF) ....all channels FIR processed.
I have found through testing that I need all the taps I can get out of the OpenDRC 2x2 units for the sub, and for the mid section.
So, one DRC-DA for the pair of subs, one DRC-DA for pair of mids. (I REALLY wish the 2x2 plugin would allow for stereo input summing, for a mono sub output .... with a 12,288 tap count...but I guess that is another thread....)
A DA-8 can handle the pairs of HF and VHF drivers easy enough.

So three miniDSP boxes.....
2 DA's with AES/coax/optical digital inputs,
and one DA-8 with coax digital input only.

What do you think is the best way to feed these three boxes?
My source is AES, but I do have a AES to SPDIF converter..
I also use a ministeamer for USB to coax..

A one-in, three out SPDIF coax splitter would obviously work, but I am having a hell of a time finding one..

I've learned I can add a DI input card to the DA-8, but that will require some custom fitting.....
and then I still need to find a one-in three out AES splitter ....which is also looking difficult...

Recommendations greatly appreciated...this seems like a trivial problem, but I sure haven't figured it out yet !

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Pls help with 3 box design... 5 years 1 month ago #26754

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@ captgnarly

Quite a setup here! No compromise it looks.. :-)
- We've never tried but splitting AES-EBU in 3 without a spliter could work or not.. never tried.. :-( You'd then have one adaptor for the DA8 (AES-EBU, like the neutrik inline one) could be one way to go I guess? .. (www.neutrik.com/en/accessories/aes-ebu-impedance-transformers/).

Having 3 x units to configure your 3way setup isn't going to be too easy as you know.. But I guess that it's a one time setup..
Have you thought of getting a DDRC-88A instead? This is analog in/out but an option I guess.. Running the BM you can have Room correction + active speaker for a similar price, but you actually have a much more powerful solution (Dirac live is much more powerful than simple FIR to our mind.. :-) Worth thinking of it..

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Pls help with 3 box design... 5 years 1 month ago #26775

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@ captgnarly

Quite a setup here! No compromise it looks.. :-)
- We've never tried but splitting AES-EBU in 3 without a spliter could work or not.. never tried.. :-( You'd then have one adaptor for the DA8 (AES-EBU, like the neutrik inline one) could be one way to go I guess? .. (www.neutrik.com/en/accessories/aes-ebu-impedance-transformers/).

Having 3 x units to configure your 3way setup isn't going to be too easy as you know.. But I guess that it's a one time setup..
Have you thought of getting a DDRC-88A instead? This is analog in/out but an option I guess.. Running the BM you can have Room correction + active speaker for a similar price, but you actually have a much more powerful solution (Dirac live is much more powerful than simple FIR to our mind.. :-) Worth thinking of it..

DevTeam


Thank you DevTeam,

I would consider the DDRC-88A, and am very intrigued by the mixed-phase filters.
But the 3 box setup is for a portable PA rig ( hopefully a super high sound quality PA rig :).
So I'm really not interested in room correction.
And I have to be able to do driver-by-driver, passband-by-passband tuning, which doesn't seem possible with the 88A.

So, still looking for splitters....
I have tried very short Y cables for both AES and SPDIF, for a 2 way split and all seems ok.
I've just read enough warnings about transmission line impedance issues that I haven't tried double Y-ing yet, to get 3 outs...maybe I will...
Thanks for the neutrik link. I do have this guy ..www.bhphotovideo.com/bnh/controller/home...7682%2C&A=details&Q=
If I can find a SPDIF splitter, I'll just split the output of this device.
If I can find an AES splitter, I'll probably add a DI card to the DA-8 to get an AES input for it.

So far, working with 2 units for a 4-way mono setup has been pretty easy, so I'm thinking no problem doing 3 units for stereo 4-way.
(Biggest help would be FIR filename recall for the 2x2 plug-in, some have requested :)

Best, Mark

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Pls help with 3 box design... 5 years 1 month ago #26778

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In case any one else looksfor a splitter ...

Found this 1-in 4-out spdif coax unit www.hdtvsupply.com/digital-audio-distribution-amp.html

Needed to change search from splitter to distribution amp.....

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Pls help with 3 box design... 5 years 1 month ago #26933

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@captgnarly
perhaps you could have glance at the www.music-group.com/Categories/Behringer...aces/SRC2496/p/P0141
providing AES-EBU digital and analog input, and the 3 digital output.

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Pls help with 3 box design... 5 years 1 month ago #26936

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Thx planetti,

I looked at that unit. It does have 3 types of digital outputs, but it's one output of each of the 3 types, .......AES, coax, and digital.
I've needed either 3 AES outputs, or 3 coax...

Hopefully, the coax box that is due to arrive later this week will feed the 3 openDRC's in parallel.
Plan on ending up with a 4-way stereo setup.....sub 6144, mid 6144, HF 2048, VHF 2048,.... taps per side....
Obviously don't need so many taps up high, but mid is about right (100-650hz), and sub could use even more...

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Pls help with 3 box design... 5 years 1 month ago #27120

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Hi Captgnarly,

If it's an AES/EBU source you have, the simplest passive solution is to split 1 in > 3 out AES/EBU using a box like this...



Sonifex CM-AESX3 costs about £105 typically .

A more ambitious option, but with many extra advantages, is the Apogee PSX100 converter, which can also do 1 in > 3 out AES/EBU.

They were discontinued many years ago now, but you can still find one on ebay for £350 - £400 if you're patient searching.

I own one of these myself, for 17 years now, and I'm using it for exactly this task, as a digital front-end to feed my three OpenDRC boxes simultaneously from my source AES/EBU signal, same as you, for my setyup with 3-way stereo tri-amped loudspeakers with full FIR correction per driver. It's working great.
Apogee PSX100 can be set to give you three AES/EBU outputs simultaneously from any stereo input path. (AES/EBU or S/PDIF coaxial or S/PDIF optical or ADAT optical channel pairs or analogue left / right balanced XLR) plus the extra benefit of front panel LED dB level metering and mute buttons, and de-jittering the digital signal down to 22 picoseconds jitter.
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Pls help with 3 box design... 5 years 1 month ago #27152

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Hi Richard, thank you.

You present some good options.

I'm running 2 OpenDRC DA and 1 OpenDRC DA-8, so 2 AES inputs and 1 spdif coax, ugh.
(Reason is because my system is stereo 4-way).
Right now, I'm converting my AES source into spdif coax and then feeding all three OpenDRC boxes with the coax splitter linked in previous post.
Seems to be working well, but I haven't had the time to measure impulse response yet.....

The PSX-100 you showed looks very interesting. Hard to believe such an old piece of digital gear still commands the price it does !
Studying the manual now......thx again

mark

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Pls help with 3 box design... 5 years 3 weeks ago #27415

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Hey Richard, I found an Apogee PSX100 converter for a little less than $400.
Seems like a bit of a digital swiss army knife .

The clincher that made me spring for it, was the 2 AES outs for the 2 openDRC DA's, and the coax spdif out for the DA-8.
No further devices needed....
All three of the miniDSP's analog outs (8 channels in my config) go into a X-32 rack.
A bit over the top for sure, but for a total of about $2500, I think I've assembled an extremely powerful and flexible loudspeaker management system .....that can't be touched for many multiples of the price....
With the Apogee unit, I also have the 3rd AES output to go straight to the X-32 and bypass all miniDSP processing.
I guess you can tell I'm psyched :) thx again, mark
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Pls help with 3 box design... 5 years 2 weeks ago #27492

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Hi Captgnarly,

Well hey, I'm glad you like the Apogee. Surprised you bought one so soon already, it's still an expensive purchase.
There's two "main" simultaneous AES/EBU outputs (at 48kHz or both required for 96kHz) plus the third Aux AES output which is always 48kHz, and you sometimes have to press and hold DIG COPY button to get it to allow pass through from one given digital format input to the same digital output formats, but it's all in the user manual.
www.apogeedigital.com/pdf/psx100_usersguide.pdf
You can also switch the output polarity (invert phase) with the DIP switch #3 at the rear, and also raise or lower the analogue input or output levels by about 10dB with the other DIP switches #4 and #5 if your signal is too strong or too weak for consumer vs pro equipment.
If it's a second hand model, you might need to recalibrate the mini-screw thingy's on the front which fine-trim the levels????

If you ever get stuck, the "factory reset" is by holding OUTPUT RES button while you switch power on, which you might need once in a blue moon.
I've had mine for 17 years and it's great, and I use it as a 1 in 3 out digital AES/EBU distribution box among other things.

One thing I've found, the MUTE buttons on the front will mute the INPUTS only (ie. what's going to AES 1 and AES 2 outputs) which may well be your digital input source, but won't necessarily mute the Aux AES output because that's an uninterupted digital through, so the danger is that if your music digital source (iPod digital dock, CD player, tablet, laptop computer, X32 digital mixer or whatever) were to glitch or drop the signal, or switch sample rates like 44.1kHz to 48kHz you might get a nasty digital glitch pop noise amplified through your speakers!!! so you want to be able to quickly use those Apogee MUTE buttons to protect your delicate tweeters and dome midranges, by ensuring they run off the two main AES output paths and perhaps make sure the third Aux AES output path - which doesn't have mute button action - is for something more robust like the woofer which can survive a digital glitch pop, if it occurs.
Given the OpenDRC's ASRC clocks are independent they will keep going and doing the LPF on the woofer anyway, which will filter the worst treble of any digital glitch out anyway going into the woofer, but you'd still hear a bassy thump.
My tip is basically, don't feed the tweeters from the Aux AES out - just in case. Use one of the main AES for delicate tweeters because if something goes funny, you can panic and press the mute buttons on Apogee to save them. My Pure i20 iPod dock sometimes does glitch like that if I wiggle the cable too much. The Apogee itself however is rock solid and won't drop it's signal but it can't be accountable for what you feed into it.

>" I also have the 3rd AES output to go straight to the X-32 and bypass all miniDSP processing."

I don't understand how you could do that??
I have a Behringer X32 mixer myself.
It doesn't have XLR style AES/EBU digital input as such (only stereo AES/EBU output or AES50 i/o format), but I do have 32ch Adat optical inputs to X32 via Behringer's optional expansion card. The Apogee PSX100 outputs all its digital outs simultaneously, so you could still feed Adat optical from Apogee into X32 mixer as well as have the three AES outs going elsewhere to MiniDSP.

Hope it all goes well,
Best regards,
Richard

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Pls help with 3 box design... 5 years 2 weeks ago #27510

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Hi Richard, thanks for the ongoing advice.

When I got the unit I checked/set the DIP switches, and did a factory reset to get 'synced' with the user guide.

Thought I had it down, and still,..... I almost returned it thinking AES1 was defective.
Until I read the manual closer :blush: ...
...and read that you have to have hold dig copy in until it blinks to unmute AES1,

I will definitely follow your advice re tweeters on AES aux. I do have some protection amp's limiters I'm using, qsc pld's, but why not minimizing testing them. A calibration check is surely due....but I can already tell it's pretty close still.

You are of course absolutely correct that the x-32 won't take an AES input.
I had strapped a Hosa AES to spdif converter on top on the x-32 rack, and after months of use had simply come to think of it as part of the mixer... i can get in brain dead grooves haha
I've never played with ADAT. Isn't it optical only?
I use the x-32's USB card to interface with a JRiver ID for music playback, and with computer for measuring (smaartlive, REW, etc)
No other uses for the USB card really, but I'm under the impression I'd lose these with ADAT unless I go through more conversion loops????

Hopefully, I'm now in the position to do some serious FIR filter comparisons. I'd like to learn the real world trade-offs between FIR and IIR, particularly for live sound, latency vs sound quality.

Thanks and very best,
Mark

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Pls help with 3 box design... 5 years 2 weeks ago #27512

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Hi Captgnarly,

X32's optional Adat optical digital I/O board would indeed replace the USB audio I/O board, but you can always swap these boards back again easily for different setups, or buy a MiniDSP USBstreamer Box to convert between ADAT and USB externally, and thus feed your computer's USB audio to and from the X32's Adat optical card via that little box alongside any other digital inputs on Adat format, without needing to swap the expansion cards. I've got one, it does the job fine.

Re: FIR latency.
Most of the time with OpenDRC running its 6144 taps at 48kHz sample rate, your 6144 tap = 128 millisecond FIR impulse is probably aligned to the main peak in the dead centre to give the linear phase symmetry of the impulse skirts either side, so you usually get 64 milliseconds of latency (ie. half the total FIR impulse duration) plus maybe negligible amounts from D/A converter delay, ASRC delay and stuff like that, but 64ms is the main number to remember.
(Obviously you can generate any custom FIR with anything from 0ms to 128ms delay using OpenDRC impulse files. For instance, if you wanted 25ms latency, you could offset your FIR impulse so the main peak was only at 25ms. RePhase lets you do that easily.)
In any case, the real overall latency you hear greatly depends on physical distance sound travels acoustically from speakers to your ears, so your listening distance in your room. The extra 64ms due to FIR filter does add a noticeable bit more though.

That 64ms latency is equivalent to sound travelling about 22 metres or 72 feet, so just imagine your loudspeakers are 22 metres further back away from you and it's about that much delay! It's not too bad if you're singing live through a mic, as many big theatre stages might be 22 metres or more to the speakers, so it's not uncommon to work with that kind of latency. Playing a musical instrument like piano or drums monitored with 64ms latency it might take a bit of getting used to, as you don't feel quite so "in the pocket" as usual but it's possible to adapt and still play okay. I've played my own piano keyboard through my 64ms latency FIR corrected speaker setup many times.

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Pls help with 3 box design... 5 years 2 weeks ago #27527

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Hi Richard,
Yes, I've come to know the openDRC 64ms 'base' well, along with a 21.3ms latency for using 2048 taps peak-centered, on the DA-8 channels at 48khz.

I haven't tried peak off-setting yet, and am unfamiliar with the sonic trade-offs in doing so.
Any good primers you know of?

I do need to work on shortening latency for any kind of live sound goal. Subs are horn loaded with another 9 odd ms of their own...so I'm well into the 70ms range....

Come to think of it, I haven't tried putting a rePhase file of anything other than 6144 taps into the openDRC....
Wonder what happens if you put in less?

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