Control Room - Direct monitoring - cue mixes

Hi all,

I apologize in advance if this question has been previously answered but I can’t find it.

I would like to know if we can use direct monitoring and control room and still listen cue mixes with inserts in headphones.

I tried everything and still can’t get the cue mix if I use direct monitoring although I read it is possible with the UR series.

e.g.
“When Steinberg launched the MR816, however, they also adapted Cubase so that, with their own hardware, the Control Room and ASIO Direct Monitoring could be used simultaneously. Cue mixes created using Studio Sends on audio tracks are auditioned with near-zero latency whatever the buffer size, and with the UR824’s DSP effects applied, just as they are in the dspMixFX utility when using another DAW”

JF Simard

Cubase 7.5.20
Windows 7

I’d like to know the answer to this as well! I’m not really sure I understand the wording in your quote … Cubase itself was adapted - how, I wonder? And how does it make sense to use a Cumix but using the hardware’s DSP effects … isn’t the Cuemix itself meant to have varying amounts of the software effects loaded up in the racks (like Aux sends on a hardware unit)?

Anyway, I’m just getting used to my UR28M, but if I were looking for another source of information on this now, I’d post in the soundonsound.com forums, in the Production forum, Forum , if you put the author’s name in the subject he may even answer it himself.

Good luck!

Bump for the weekend warriors … :slight_smile:

Tried it out … with Direct Monitoring I’m not able to have an audio file’s insert be applied /heard through the Control Room.

I’d sure like to know what this author meant as well, or how to do it, etc.!


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[EDIT]
OK, I think I figured this out, to work as the article suggests.

Here’s how I look at it (corrections welcome from anyone):


The “expected” behavior, in non-Steinberg interfaces, is that the amount of delay into your cans when tracking through the Control Room (or equivalent in another company’s interface) would be at the mercy of the interface’s buffer size/latency. So if you need a large buffer size to prevent crackle city, and are drumming along to a backing track, and everything is routed through the Control Room, you won’t hear the drum hits for a long time after you hit them, long enough to throw the timing off perhaps. Turning down the buffer size would reduce the latency, but for some recording situations it may not be able to be turned down enough without running into the limits of the interface. Direct monitoring would let you hear your drum hits instantly even with a large buffer, but in that situation the recorded audio would be out of sync with the earlier recorded backing tracks.



What I think he is saying the great advantage is in the newer Steinberg interface is:

You can have your buffer/latency incredibly large … but still have the Cue Mixes in the Control Room come out as “instantaneously” as the audio you are tracking live with direct monitoring on. And, those Cue Mixes are intact in terms of inserts and panning (by my checking), and I’d guess sends and other things too.

And wait, there’s more :mrgreen: : the live audio can have all the DSPfx (Channel Strip, Rev-X) applied through the hardware channel as you hear it in the cans - again, all in perfect sync with the backing track and its Cue Mix pans/effects etc. through the Control Room.

So, if I understand all this right, and as I tested it tonight: BIG buffer + Control Room Active + Live Audio DSPfx = No Problem!

(I’m still not sure what the author means when he says setting things up this way “kills the Control Room Metering …”).


jfsimard, you wrote: “I tried everything and still can’t get the cue mix if I use direct monitoring although I read it is possible with the UR series.” I’m trying to think about where this could go wrong, are you sure you have the Blue “Cue” button pressed (as opposed to the yellow/gold “Mix” button) in both the top and the bottom portions of the Control Room Panel? And that the Cues themselves (in the actual audio column) are “powered” on, with the blue light (not the white one), and with the green color on the slider? I always test with the Listen button, and that weird panning is heard in the Control Room to make sure I’m hearing the Control Room, not the main mix.

Hope that helps, and if anyone finds something wrong in what I’ve said, please chime in here!

Hi Alexis,

I didn’t realize there was activity on this post, I stopped looking after a while. Thank you for your input.

Since then, I got to understand more about how to work with Direct monitoring and the Control Room.

You wrote:
“(I’m still not sure what the author means when he says setting things up this way “kills the Control Room Metering …”).”

From what I experimented, is that the Control Room Meter doesn’t work anymore while in Direct Monitoring.

Does it work for you?

JF Simard

I’ve got it working so far on my UR28m I’m attaching a jpeg of my controlroom connections, to use direct monitoring, you have to connect the input(s) to the regular input page. ie “Direct Monitoring” does not work when assigned to the control room. The ques work through the channel that you are recording to, just make sure that the que to that channel is turned up.


Not sure why, but the sync is right. To get 2 ques, just replace the phones channel with que 2.

Hello Alexis and Ross and all you MR816 users ;

I am glad your asking these direct monitoring and “no and low” Headphone Cue latency questions because it seems
very few of us has figured it out and most are frustrated and stuck… I’m on a mission to create world class -“zero” or almost zero latency headphone cue mixes here inDallas TX. Seems like, “Marcus” at Norburry Brook studios in UK is the clearest on the “How to’s and Why’s” and is successful.

OK I use 2 MR 816csx’s, they are Fire Wired and linked together with SPDIF and FW as Steinberg recommends. Everything shows up as it is supposed to in DEVICES and I use the Control Room and the Headphone CUE sends very successfully with EFFECTS as it was designed to be used out to 6 satellite boxes in my 4 room studio via CAT5e. OK the 4 mono sends and 1 GLOBAL stereo in 5/6 ( in CUE cans) works fine. BUT, there is still latency and comb filtering going on in the HeadPhones. (Very fast QUAD computer, ASSUS motherboard) and latency of 3.78 at 128 buffer.

So If I can still use the Cubase Cue Sends with DIRECT Monitoring can anyone explain it to me and perhaps MAP it out. I’m close but still not successful.

Can I send via Direct Monitoring out the ADAT outputs 1 and 2 any prefader audio. I’d be satisfied even if I lost 2 channels per MR816. If I can do this I will send this “pre-fader” audio to a completely separate and stand alone digital mixer (X32core) and build my ZERO latency Headphone CUE mixes in the offline mixer and then send them out the ULTRANET port to be redistributed thru out my studio.

I track alot of rhythm sections here in Dallas and as I said above MY mission is World Class Headphone Cue mixes especially for ensemble players and vocalists.

Any Help appreciated and thank-you in advance.

Ken Boome in Dallas TX

214-321-BOOM (studio)

Hi all.
I see this is an old post, but since recently purchasing the Steinberg Ultimate Recording Package I too have run into this problem.

I have found a workaround, albeit a little messy in terms of the mixer layout but working.

I’m using the UR824. This has 4 stereo mix layers. My mission is to provide a rough mix to the performers with 6 “more me” direct monitor lines. (6 lines are fed to each aux in of a Behringer HA6000 providing the 6 headphone mixes)

When in Direct Monitoring mode, Cubase will only monitor live input to the single DAW channel on each of the UR824 mix layers.

(using the DSPmix24 app you can see the DAW channel next to the master)

Set the Line outs of the UR824 for each mix output (ie LINE OUT 1 & 2 = Mix1 L & R, LINE OUT 3 & 4 = Mix2 L & R…)

It is possible to add outputs to these in VST connections. (I see it’s called Audio Connections now)

So in there I have my main Stereo out routed to mix 1

Add 6 mono outputs and route them to the next Mix layers;

	mono out 1 > Mix 2L 

	mono out 2 > Mix 2R.... and so on. 

Now this is where it gets messy.

For each live channel you wish to record you’ll need another one for each foldback mix.

You can think of it as creating a separate mixing desk for each monitor output.

If the bassist wants more bass, create another bass channel, give it the bass input routing but set the output routing to whichever the bassist is on (in my case the bassist is plugged into channel 3 of the headphone amp, so route the channel to mono out 3 ……I’ll rename these later)

On this new Bass monitor channel, you can click the audio monitor button and it will send straight to that output assigned. Direct monitoring and latency free.

So I have the 16 tracks I wish to record. These are routed to Stereo out 1, armed and audio monitor is on. These are the channel I’ll create a rough mix with.

I have another bass track routed to mono out 3, not armed but audio monitor is on. The channel fader is used to control the send.

And one more vox channel for the vocalist on mono out 6 not armed, audio monitor on.

Looks messy, but I’m looking to find a way to group them in the visibility column.

Any thoughts and suggestions are welcome. I hope someone finds this useful.