X32 Discussion

Re: Cases for the X32

My X32 showed up yesterday. I seem to be having a compatibility problem, though. It seem this board is quite incompatible with me getting any sleep! :O Ah well, it was a lot of fun playing. Still got a few (er, a TON) of things to figure out, though.

-- Mitch

Hi Mitch,

Congrats on the new board!

The BEHRINGER forum is up and running, you may also want to check out posts and info there as well:
Forums

This is our X32 YouTube channel with some videos and past X32 Webinars:
X32 Resources - YouTube

This link is for our upcoming X32 Webinar/User Group Events:
X32 Live! Webinar/User Groups

Hope it helps!

Best
Joe Sanborn
Manager, Channel Marketing
MUSIC Group
BEHRINGER
 
Re: Cases for the X32

Joe -- thanks for all the resources. In fact, I'm on the forums, have been checking out the videos, and am signed up for the seminar on the 19th! (And looking forward to the video from the most recent webinar that I missed out on!)

Incidentally, my son and I did a whole lot more structured playing last night and made a lot of headway toward setting the board up they way I plan on using it. Love me some Xcontrol. Tonight the goal is to record into Presonus Studio One 2.

-- Mitch
 
Re: Cases for the X32

Hello,

The case that comes from multi-caisse is much more expansive, it is heavy duty made for rough transport. the latchs and the wheels are high quality, the foaming inside is custom fit. I have added some air irrigation in the bottom. its a bit heavy 100LB but still very nice case !

Othmane

I am in the process of building my own case. Nose cone, console tray with doghouse, and space underneath the console for a rackmount server (Windows7 and perfect for running console software, amp control from FOH as well as multichannel recording) and a clear-com remote station. I don't have pics just yet, but the case should only be about 5 inches taller than the Gator Case. The dog house will incorporate the two monitors as seen in the pic. Both are 20" LCD on swivel mounts.

Having it in the house for a few days has also given me the chance to set up some templates for the various ways I have been using digital console the past few years. Lots of time spent now, but hassle free when the show setup time arrives.

Oh, and the case will more than likely have a padded armrest built into the tray that accompanies the console just right, ass well as a leg system so on person can flip it up quite easily- that part is still in development. I hope to have the tray and arm rest with dog house completed by the end of next weekend...
 
Re: How to do an analog insert?

To round out the explanation above, individual physical inserts for each channel do not exist on this console. One can assign any one of the 6 aux pairs as an insert "out and back" to any channel.
gp
Thanks guys, very simple. A couple of questions though:
Does the return signal also appear on the respective Aux return channel?
And what do you do if you want to do a bunch of inserts like 16 channels of Dugan Automixer?
Mick Berg.
 
Re: How to do an analog insert?

Thanks guys, very simple. A couple of questions though:
Does the return signal also appear on the respective Aux return channel?
And what do you do if you want to do a bunch of inserts like 16 channels of Dugan Automixer?
Mick Berg.

One of the obvious limitations of the X32 is that it only has six pairs of sends and returns. Of course, there's loads of "rack gear" online, so for a lot of applications six pairs isn't a problem. But six is less than sixteen.

You could connect ten of your mics to channels 17-26. Set gain and eq, then take the direct outs to your automixer. (I'm assuming pre-fade. If post-fade, you'll want to change the numbering scheme to get everything in the same layer). From the automixer, bring the channels back in on 1-10. Use your other six mics on 11-16 with the inserts. (This way all sixteen channels are on the same layer.)
 
Re: How to do an analog insert?

Thanks guys, very simple. A couple of questions though:
Does the return signal also appear on the respective Aux return channel?
And what do you do if you want to do a bunch of inserts like 16 channels of Dugan Automixer?
Mick Berg.

I guess the X32 might not be the right mixer if you want to use 16 channels of Dougan. Now, depending on which Dougan you are planning to use, there would of course still be ways that might be satisfactory without crippling the X32s capabilities in the process. Using up nearly every input and output might be OK if you don't need the inputs and outputs for anything else, and would work, albeit with some latency, if you have a Dougan with analogue I/O. A better option for a Dougan with analog I/O might be to use outboard analogue preamps, and only enter the signal into the X32 after the Dougan, thus avouiding some latency and save some I/O on the X32. If you have a Dougan with ADAT, the best option might be to go through the Firewire interface on the X32, into a computer with an ADAT card, using the couputer as a bridge between the X32 and Dougan, this could get very complicated, but could possibly be achieved without adding too much latency. Another option would be to keep your fingers crossed for Behringer to bring out an ADAT card soon, or an insert card.
 
Re: How to do an analog insert?

I guess the X32 might not be the right mixer if you want to use 16 channels of Dougan. Now, depending on which Dougan you are planning to use, there would of course still be ways that might be satisfactory without crippling the X32s capabilities in the process. Using up nearly every input and output might be OK if you don't need the inputs and outputs for anything else, and would work, albeit with some latency, if you have a Dougan with analogue I/O. A better option for a Dougan with analog I/O might be to use outboard analogue preamps, and only enter the signal into the X32 after the Dougan, thus avouiding some latency and save some I/O on the X32. If you have a Dougan with ADAT, the best option might be to go through the Firewire interface on the X32, into a computer with an ADAT card, using the couputer as a bridge between the X32 and Dougan, this could get very complicated, but could possibly be achieved without adding too much latency. Another option would be to keep your fingers crossed for Behringer to bring out an ADAT card soon, or an insert card.

His name is Dan "Dugan" and I agree his solution sounds a little high end for this mixer. Dan sells a card to do auto-mixing in Yamaha digital consoles. Since the Dugan AM algorithm is just math, it seems a digital mixer could accomplish the basic AM function with a software tweak. I realize AM is a mostly speech, specialized application, but at some point this will be another plug-in for digital consoles.

One guy wrote a program that works off digital console meter levels to drive faders and that works mostly but it is somewhat compromised in that it can't discriminate between coherent and incoherent sources (3dB gain error). Being able to access and crunch complete audio signal data could deliver a fully effective solution.

JR
 
Re: How to do an analog insert?

I've had a search - but I'm having no luck making the USB drivers work on my Macbook pro, via parallels and windows 7. Pushing the plug in, and selecting windows simply shuts windows straight down. It will not reboot with the plug in, but is happy without - but again fails on any attempt to load the driver. Any clues folks? The Mac driver works fine, but I have Cubase on the windows installation and wanted to try this.
 
Re: How to do an analog insert?

Hey guys,

I have a few questions about the X32 regarding recording. I'm currently using it with Cubase 5 via USB.

I've set up a Cubase project with the purpose of recording all 32 tracks and sending them right back to the X32 so the X32 can do all processing (EQ, Comp, Gate, effects, etc). In the X32 Setup I've set up the input source for 1-32 to be the "Card" inputs and set up Cubase to output 32 tracks to the X32.

Question: When I've set up the 32 inputs on the X32 to be the Card inputs, is the only way to monitor the inputs to toggle the "input monitoring" button in Cubase (per channel)? I don't get any signals coming through the board unless I turn the input monitoring on. The problem I'm having is that if I toggle input monitoring for all 32 inputs, I'm getting a significant amount of crackling and popping which I assume is caused by a processing overload. I've increased the latency but I cannot record with the increased latency; it is just too much.

Note: Currently, I have set up 1 scene where the input channels are local to the X32 and another scene where all channels go through the Card. When recording, I use scene 1 and for playback from Cubase I use scene 2. This is a bit of a hassle. Question: Is there a way to get around this? Ideally, I want to record and monitor with 0 latency.

I would like to record with the Cubase metronome. To have the metronome play back through the X32, I have to select at least 1 of the 32 outputs in the VST Connections page in Cubase. The problem comes when I have the 2 scenes as mentioned before. If I have the metronome to play back on Track 30 for example, on scene 1 it will play back on the AUX 1 channel and on scene 2, it will play back on Track 30. This is frustrating. Question: Is there a way to have the metronome output to AUX1 on both scenes even though I'm using all 32 outputs already?

I hope I've made enough sense for y'all to help me out!

Thanks in advance,
Paul
 
Re: How to do an analog insert?

One guy wrote a program that works off digital console meter levels to drive faders and that works mostly but it is somewhat compromised in that it can't discriminate between coherent and incoherent sources (3dB gain error). Being able to access and crunch complete audio signal data could deliver a fully effective solution.

Latency of the desk in generating and sending out meter information was also a ploblem that killed off that programmer's effort.

I can see three opportunities to get an auto mix solution working.

1. Read the osc meter parameters and send back osc to control levels.
2. Analyse channels (on laptop via USB out) and send back osc for control
3. Analyse channels (on laptop via USB) and send back audio via USB (an auto mixer insert)
4. Put enough requests into Behringer that they do it. I would suggest a custom auto mix VCA group that channels can be added to that attenuates the channels independently and automatically but also uses the fader as a master. Channels can be added and subtracted as is done for every other vca.

If I get the time I might experiment with options 1 and 2.

Alan
 
Re: How to do an analog insert?

Latency of the desk in generating and sending out meter information was also a ploblem that killed off that programmer's effort.

I can see three opportunities to get an auto mix solution working.

1. Read the osc meter parameters and send back osc to control levels.
2. Analyse channels (on laptop via USB out) and send back osc for control
3. Analyse channels (on laptop via USB) and send back audio via USB (an auto mixer insert)
4. Put enough requests into Behringer that they do it. I would suggest a custom auto mix VCA group that channels can be added to that attenuates the channels independently and automatically but also uses the fader as a master. Channels can be added and subtracted as is done for every other vca.

If I get the time I might experiment with options 1 and 2.

Alan

I don't know what an osc meter parameter is.

Latency is not a huge issue since gain moves are generally smoothed, but the first MSecs of a new talker could be chopped off if latency is significant, clicks if you try to open up the channel too fast after that lag. This could be mitigated by limiting the channel attenuation to only as much as needed for effective feedback control instead of full cut reflecting actual input levels. If the channel is only dimmed a few dB, start up is only a little soft, not completely trashed.

Again the general flaw with working from already rectified signal levels is lack of coherency information. AM math requires a dummy sum of the raw audio signals to use to compare to level of individual stems for gain apportionment. Pretty simple, but a powerful technique when many open mics are involved. Built into a digital mixer, the AM gain could be layered on top of normal fader moves.

Sorry for the veer... not a huge application for musical sound reinforcement.

If you can pull actual channel audio into a remote PC and sum them there (or set up a dummy bus in the X32). You could then perform the Dugan gain sharing based on that AM sum. You could calculate fader moves to send back. Dedicating a dummy bus for the AM sum inside the X32 corrects for the coherent-incoherent issue, leaving only the attack time as a potential execution issue working from meter levels. Drilling down into the raw data eliminates even that compromise. [/veer]

JR
 
Re: How to do an analog insert?

I would like to record with the Cubase metronome. To have the metronome play back through the X32, I have to select at least 1 of the 32 outputs in the VST Connections page in Cubase. The problem comes when I have the 2 scenes as mentioned before. If I have the metronome to play back on Track 30 for example, on scene 1 it will play back on the AUX 1 channel and on scene 2, it will play back on Track 30. This is frustrating. Question: Is there a way to have the metronome output to AUX1 on both scenes even though I'm using all 32 outputs already?
Given that the metronome sound quality is of no great importance, you could output the metronome thru the computer soundcard and cable it to one of the AUX inputs. There will be some latency, but that can easily be compensated for(having a reference click track for the recorded material, and one advance click track that makes the round trip to align with the reference track)

As for input monitoring, why bother except for initially verifying that recording is ok? Once everything is set up and relative levels between the daw and the X32 are set, you are better off monitoring straight thru the desk while recording. When you are dubbing, something that might not be a great idea with usb, you need to align the new tracks post recording anyway, so monitoring a mix of playback tracks and live via the daw is just going to be messy at the best of times.
Remember the old multitrack reel-to-reel machines, one would use the recording head for dubbing playback, and only use the dedicated playback head for mixdown, a compromize, but a lot less messy than having offset tracks and realigning the heads for each take. Actually, not monitoring through the daw is probably not a compromize at all, because I can't see what is to be gained by doing it. In the old days a tech might monitor the playback heads to keep tabs on the saturation etc. but that tech would not be concerned with any timing issues between live and playback, because he would only listen to playback and be concerned with the sound, not the timing and musicality of the take. Recording to daw, there shouldn't be any considerations regarding the quality of the actual recording process, and thus no need to monitor.
 
Re: Automixing

His name is Dan "Dugan" and I agree his solution sounds a little high end for this mixer. Dan sells a card to do auto-mixing in Yamaha digital consoles. Since the Dugan AM algorithm is just math, it seems a digital mixer could accomplish the basic AM function with a software tweak. I realize AM is a mostly speech, specialized application, but at some point this will be another plug-in for digital consoles.

One guy wrote a program that works off digital console meter levels to drive faders and that works mostly but it is somewhat compromised in that it can't discriminate between coherent and incoherent sources (3dB gain error). Being able to access and crunch complete audio signal data could deliver a fully effective solution.

JR
Hmmm....sorry for the misspelling. Given that sources for automixing would be basically incoherent, wouldn't it be reasonably safe to calculate gains based on that assumption, or would that result in some instability due to random or cyclic coherence between the sources?

Latency is not a huge issue since gain moves are generally smoothed, but the first MSecs of a new talker could be chopped off if latency is significant, clicks if you try to open up the channel too fast after that lag. This could be mitigated by limiting the channel attenuation to only as much as needed for effective feedback control instead of full cut reflecting actual input levels. If the channel is only dimmed a few dB, start up is only a little soft, not completely trashed.
I guess it would be quite viable to combine a (slowish) external gain control with the somewhat faster onboard gates, thus not having to involve the external processing in anything but the actual level riding, not ever having to close down a channel further than the feedback safe level and still have no noise issues because the gates handles that.
Now, forgive me for being absolutely ignorant, but what kind of live automixing scenario is there that can't be solved by gating and compressing individual channels plus groups. While I certainly understand the difference between compressing and automatic gain riding, once you set the compressor to quite long times ( the X32 compressor can be quite sluggish at 120mS attack, 2 S hold and 4 S release), and for the play we did last week-end, I found it quite usefull to just replicate what I've done before with the actors earset mikes being open all the time and just gating and soft-knee comping. Given the extra capabilities now at hand, I'm thinking maybe expanding at the input channels and compressing the group channels to get closer to a level control of sorts, or would that be counter-productive?
 
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Re: How to do an analog insert?

Hmmm....sorry for the misspelling. Given that sources for automixing would be basically incoherent, wouldn't it be reasonably safe to calculate gains based on that assumption, or would that result in some instability due to random or cyclic coherence between the sources?
Not to continue this veer but it matters quite a bit for accuracy of gain sharing under different source conditions. A distant sound source that is picked up equally by multiple mics, or say from one talker equidistant from two mics would be roughly coherent and combine linearly, so gain calculation assuming incoherent would be incorrect.

One powerful application for AM is reducing feedback from a room monitor speaker when many open mics are involved. Imagine a monitor speaker sound signal being picked up by 10 open mics, worst case if this speaker is equidistant from each mic, they could sum to roughly 10x one mic (not exactly), but if ASSumed to be incoherent sources for gain sharing, the per channel gain would only be reduced by SQRT 10, resulting in less dB margin before feedback occurs.
I guess it would be quite viable to combine a (slowish) external gain control with the somewhat faster onboard gates, thus not having to involve the external processing in anything but the actual level riding, not ever having to close down a channel further than the feedback safe level and still have no noise issues because the gates handles that.
Now, forgive me for being absolutely ignorant, but what kind of live automixing scenario is there that can't be solved by gating and compressing individual channels plus groups. While I certainly understand the difference between compressing and automatic gain riding, once you set the compressor to quite long times ( the X32 compressor can be quite sluggish at 120mS attack, 2 S hold and 4 S release), and for the play we did last week-end, I found it quite usefull to just replicate what I've done before with the actors earset mikes being open all the time and just gating and soft-knee comping. Given the extra capabilities now at hand, I'm thinking maybe expanding at the input channels and compressing the group channels to get closer to a level control of sorts, or would that be counter-productive?

AM is a premium solution for managing a large number of open mics. The Dugan algorithm portions out the gain per channel so the total gain in all channels combined is equal to one mic at full gain. You could do this manually by riding faders, but not easily. Gating does not resolve the scenario when multiple channels inputs are above threshold so open up.. Enough people talking at the same time could start feedback, and the gates are not going to close once the feedback starts. If the system doesn't feedback with all the channels open you probably don't need AM.

Again I apologize for veer.

JR

PS: For TMI years ago I did design a cheap NOM AM input module solution for install mixer-amp products, where each mic input had a gate threshold, and when that channel turned on, it dimmed all the other channels a fixed amount. Cheap and crude, but rough NOM correction for use in a budget install product.
 
Re: How to do an analog insert?

Behringer folks: Would it be possible to make it so the X32 auto-stops recording and saves when the 2GB limit is reached? There have been a couple instances where I have been recording a show and gotten busy, forgotten to hit "stop" and the resulting file is 0 bytes, a blank file and a lost recording.
 
Re: How to do an analog insert?

Behringer folks: Would it be possible to make it so the X32 auto-stops recording and saves when the 2GB limit is reached? There have been a couple instances where I have been recording a show and gotten busy, forgotten to hit "stop" and the resulting file is 0 bytes, a blank file and a lost recording.

Are you on firmware version 1.09? According to the release notes this has already been taken care of.
http://www.behringer.com/assets/X32_Release_Notes_1.09.pdf
 
Re: How to do an analog insert?

That's interesting. I am using the newest firmware. This makes me wonder what caused the 0 bytes file I ended up with from my last recording.

Removal of the USB stick or powering down before the end of file marker is recorded is the usual culprit, but I don't know enough about how Behringer does the file handling to claim this as certain causality.
 
Re: How to do an analog insert?

Greetings ..
I am following this thread, and I have also had many problems in the recording studio environment .. Noise clips, panel buffer incomprehensible, complex routing configuration ..
Etc etc. .. I'm concluding that the X32 is not a decent mixer for recording ..
Now he has a new problem .. as recording MIDI from an external keyboard or other MIDI instrument???
X32 is a live sound mixer
X32 NOT used for recording studio:(~:-(~:sad: