Pre-Fader Direct Outs

cjac9chris

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DM-3200
Hi everyone! I'm new to this forum as I just bought a DM-3200 - this is my first post.

I did a search and didn't find a thread covering this topic. Forgive me if there is one and I missed it. Here's what I can't seem to figure out...

I'm using the Behringer Powerplay P-16i (think Aviom) and would like the Behringer mixers to see pre-fader direct outputs from the DM-3200. That way when I'm tracking a band I can put together my mix and not affect the individual Behringer mixes.

I've tried:

1.) Sending Buss/Direct outs out of ADATs but when I move any faders it affects the level going to the Behringer.

2.) Using pre-fader insert points to send to the Behringer. The audio to the Behringer is unaffected BUT then no audio passes out of the channel on the DM-3200 because it isn't returning after being sent out of the ADAT.
 
Easy. First hook up adat out from your dm to the adat into the Berhinger. (8 channels only)
Now go to the DM keypad and go to routing/output tab. Use POD4 (right side) to select M/L under "input Bypass" Now use POD3 to select ADAT. Now look at the left side under ADAT. Use your arrow keys and jog wheel to select M/L 1 for spot 1, hit enter, M/L 2 for spot 2, hit enter, etc. until you have all eight ADAT outs setup. This will allow your DM mic input signals to split. One set goes to your channel faders on the board, and another set goes unaffected out via the ADAT outs. For 16 channels, you will need two ADAT outs. You could use your assign sends for 4 more outputs as well.
 
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Off topic: How do you like the P16 so far? I'm thinking about getting it as well.
 
@TascMan - that did the trick! I was testing everything out with music through the 2-track in and apparently that's the only input you can't split like that! Works great with TDIF though!

@henkmeid - Well, now that I've got it working properly... it's pretty cool! It feels way more solid than the Aviom system and sounds a lot better than I remember those units sounding (they can sound pretty harsh and in the wrong hands, phasey.)

I think for a digital mixer based setup you can't go wrong with it, esp. for the price. With all the units I think it's about half of an Aviom system and you can't put a price on not having to deal with headphone mixes in my book. Happy players play better and engineers that don't have to deal with constant "more me" requests can focus on getting great sounds and getting a really good rough mix together.

The only con that I've found is that there is a 3-band EQ w/ semi-parametric mids and adjustable limiter built in. Plenty of room for user error. It'd be great if there was a pre-set brickwall limiter just for surprises to protect people's ears and maybe a master EQ to adjust for different headphone models but this thing allows you to treat the limiter like a stereo buss compressor threshold control and you can EQ every channel (which it then saves and you can't see what is saved because there's only one set of controls and they aren't recallable.)

Haven't gigged with it yet... but I have high hopes as long as players don't go messing with the EQ and Limiter!
 
Another question. Maybe another topic or maybe I'm just plain stupid.

But using this combination with adat. Am I still able to track to my DAW in 96k? Because adat is 44.1. Or will there be some sort of conversion in the desk?

Or is is best to use a analog card to connect to the p16? You mentioned tdif as well. Is that able to carry 96k?
 
Not a dumb question at all. To be able to transfer all 8 channels of audio via ADAT, you are correct; you must be at a sample rate of either 44.1Khz or 48Khz. This is a limitation of this protocol. However you CAN transfer audio over ADAT at 88.2 KHz and 96Khz using the SMux protocol, which the DM boards support. When doing this, you lose one channel of ADAT for every channel you want to transfer. So, instead of 8 channels of audio, you are limited to 4 channels when you are at 88.2 or 96. This will happen automatically so don't be surprised. I have mine set up this way since I record at 88.2. This is also the case for the TDIF ports in the DMs. 24 channels become 12, when using the upper sample rates.
 
Thanks for that clarification!

The only downside is, i just found out, the p16 only support 44.1/48k ADAT and no SMux. So i guess it will be analog anyway. Together with the DM4800 you get 8 assignable sends and add 1 analog card and you have 16 analog outs.

Kind of a bummer tho, saves a lot of cabling.
 
Just been checking the manual for a bit:

Digital inputs
The word length (16BIT or 24BIT) and the input selected using the routing screens (XLR or RCA) are automatically determined and displayed.
Also, in the case of a “double-speed” input (that is,88.2kHz or 96kHz), the display shows HISPEED, otherwise it shows NORMAL.

The DM-4800 includes a frequency converter, which
is capable of taking an input frequency of 44.1, 48,
88.2 or 96kHz (±6%) and converting it to the project
frequency. Use the on-screen FS CONVERT button to
turn the converter on and off.

NOTE
The signal from any input used with a sampling fre-
quency converter cannot be used as a sync clock source
(see “Clock setting” on page 56).

But this says it about the input frequency to be converted, will this work with output as well? Taking a 96k from the board en put it thru adat with 44.1/48k?
 
Not sure about output, but this part of the manual is about the digital ins/outs, and AFAIK not about the ADAT or TDIF outputs..
 
rbrezins said:
Sample rate converter is only active on the Digital IN 1 and 2 (XLR/RCA) inputs.
This converter is not available at any digital output.

Exactly! This is the reason I have had to employ two sample rate converters for my D1 and D2 digital outs. I have to resample down to 44.1, go through the sample rate limited equipment, then have the DM resample the inputs back up to 88.2 for the project.

I have used, and occasionally still do, a MOTU 2408. It has 8 channel of analog I/O, 24 channels of ADAT and TDIF I/O, available at the lower rates and 12 channels at the higher rates. I AM able to convert an analog signal to 88.2 in that and send that signal out over ADAT, just 4 channels though, as it utilizes the SMux protocol. Again, same for the TDIF which is also available in the 2408.
 
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TascMan said:
So, instead of 8 channels of audio, you are limited to 4 channels when you are at 88.2 or 96. ........ This is also the case for the TDIF ports in the DMs. 24 channels become 12, when using the upper sample rates.

Is that right? I thought I read somewhere that TDIF offers 24 I/O regardless of sample rate - couldn't find it in the manual though..... :?:
 
TDIF is under the same constraints as ADAT for high sample rate digital transfer. Sample rates at 88.2 to 96Khz require halving the number of channels in order to handle the higher rates.
This is one of the FAQ about the MOTU 2408 MK3: http://www.motu.com/products/pciaudio/2408/faq.html 4th question down. This is not a 2408 limitation, it's a standard digital protocol limitation.

Here's a wiki on the subject. Look down at the bottom regarding Version 2.0 and the "reduced channel count" http://en.wikipedia.org/wiki/TASCAM_Digital_Interface

Tascam's X48 might have been the only unit to use a version of the TDIF protocol at full channel count, but the DMs do not.
Another reference:
http://www.motunation.com/forum/viewtop ... =2&t=36077
 
TascMan said:
Tascam's X48 might have been the only unit to use a version of the TDIF protocol at full channel count, but the DMs do not.

Aha - that's where I read it - I had an x48 for a while :idea:
 
Tascman,

I believe the approach you describe is the "Input Bypass" approach. What about the Direct Out approach? If I set my ADAT1 output to BUS1/DIRECT1 on a DM4800, and I then select DIRECT for Channel 1 assignment, I'm supposed to send pre-fader into ADAT1 right?
Is this an alternate solution?
My goal is zero latency drum recording, so I want input1 to go to ADAT1 pre-fader, while adding DM4800 post-fader EFX for the drummer's headphone monitors during tracking. I don't like the Input Bypass approach because it precludes me from using my ADAT outs from the bus without de-assigning the Input bypass.
 
I had to reread this thread, as it's almost a year old.
Philippe...Hi,
I understand your question all the way up to the last line. I am not sure what you mean there. No matter, read on...
If I set my ADAT1 output to BUS1/DIRECT1 on a DM4800, and I then select DIRECT for Channel 1 assignment, I'm supposed to send pre-fader into ADAT1 right?
No.
Just to be clear about your first question, the term "DIRECT" does NOT actually mean DIRECT to an output, pre-module (Pre fader). It means DIRECT to any destination, POST module. When you press the DIRECT button, you first have to associate a module to it, not an input signal. This tells the DM where you are going to send the output of the module...in this case, out Direct (to BUSS CH 1), which will then be POST module. It's a bit misleading, I know.

So, the first thing to realize is that the only way to have a signal effected (via effects and/or volume-pan) is to send it through a module (channel input) and then assign that to any buss or direct output...Stereo buss (most common), Aux Buss, or any of the 16 channel Busses (24 for the 4800), then out from there. Therefore, the only way to send a signal somewhere without going through a module is to assign that signal out of the board via "Input Bypass" (better read as "Module Bypass") The OP was looking for a way to send his signals directly out of the board via ADAT without having them effected by any module. The best way (I could think of) is to utilize "Input Bypass".

On to your issue,
You would like to record your drums with zero latency. So, the question would be, where are you experiencing latency? In your headphones/Monitors? Are you monitoring through the DM or are you listening to the playback of the individual DAW channels as they are being recorded? I ask that because, typically, you should be monitoring your drums using your channel modules going out either your Stereo Buss or ASSN send channels, while the signal itself is being sent off to your DAW unaffected by volume/pan or effects by utilizing "Input Bypass".
My goal is zero latency drum recording, so I want input1 to go to ADAT1 pre-fader, while adding DM4800 post-fader EFX for the drummer's headphone monitors during tracking. I don't like the Input Bypass approach because it precludes me from using my ADAT outs from the bus without de-assigning the Input bypass.
If you want INPUT 1 to go out ADAT1, pre fader (pre-module), then you would need to use "Input Bypass". (Think Module Bypass) as described above from the OP's similar question. That will take care of that.
Next, you want your drummer's headphones to hear a proper mix of himself, and anyone else in the room, complete with the DM's effects mixed in.....There is a great way to do this and it involves the AUX busses and a pair of ASSN sends. Since this is getting a bit long, I'll keep it short. Basically you need assign Aux 5/6 SEND to Effect 1, and AUX 7/8 to Effect 2. Then bring Effect 1 and 2 RTNs back into two paired stereo modules, say channels 27/28 and 29/30 (it really doesn't matter) and run those faders up to unity. Then assign those modules to the same buss (like the stereo buss) that his headphones are assigned to. Now, when you bring up the associated encoder, a parallel version of the module's signal will go out to the effect, and you will hear the return. It works great.
 
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Thanks a lot for revisiting.
Even after your explanation, I don't understand what Direct is useful for, but I don't think I need it. I will use Input Bypass especially because I just ordered a Firewire card for the DM4800 which gives me 32 inputs and outputs into ProTools so I don't mind to permanently assign 24 of them to "fixed" mixer inputs. Still leaves me 8 inputs to be more flexibly allocated to 8 bus slots.
On your last point about drummer mix, yes the latency is the stereo return from ProTools after mixed "in the box", as my PC needs at least 256 samples, which creates audible latency. My goal is precisely to create separate "live" mixes for the drummer and the guitarist with wet module signals (EQ/Comp/etc) but sending "dry" bypassed signal to Protools and remix drums in PT later with better EFX.
If I understand correctly, what you describe is a way to use two built-in FX and mix them with the dry drum mix right? I am already using the two stereo FX that way.
Is the best way to have a different mix for drummer versus guitarist to use say Aux1-2 assigned to Assignable send 1-2 (via 2 bus channels) itself going into a headphone amp stereo input 1, and likewise use Aux3-4 for a second mix and send to another input 2 of my multi headphone amp?
 
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Here is a paste from the 3200 manual about DIRECT assignments....(same for both 3200 and 4800)

"When a channel has a direct output selected, the
direct output from that channel overrides the correspondingly-
numbered buss."

The 3200 has 16 assignable Buss channels, (24 for the 4800) and 32 full action (for lack of a better term) modules. (48 for the 4800) In the routing page, you can assign the DIRECT output from any of those modules to any destination, without having to go through a buss. (Buss channels, stereo buss, aux buss, etc.) It's the fact that you can bypass buss assignments that make the DIRECT routing option useful. But, as stated earlier, these signals are now POST module.

"The direct option allows the selection of only the
direct module outputs (1–32) as sources. Note that
the selection of DIRECT as a channel destination,
using the selection keys, removes the channel from
the busses."

By using the DIRECT method, you are taking the output of a chosen module and sending that to a destination. Since you are using that module as a source, your signal will be post effects/fader/pan, etc. Typically though, you do not need to use DIRECT, unless you want to send your signal POST module out to FW or some other destination. You don't need to select DIRECT if you want to use the STEREO buss, since it IS it's own buss. I haven't really needed to use the DIRECT function for anything at this point.
It is because of this that many of us are using INPUT BYPASS to sort of "split" the signal right after it comes into the board. One copy of the signal goes to your daw channel input, usually via FW, and the other signal goes to a channel input module, to be mixed, and sent out either a cue mix, or the stereo buss for your Control Room, or anywhere else you want to send it. The advantage of Input Bypass is that the original signal is unaffected by the module changes you will need to make as you create a mix for your cue and/or Control Room
 
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Thanks, now in understand, it's just a way to bypass a layer of buss piping.
I don't need that, but I will definitely use the Input Bypass as soon as I receive my 32 in/out Firewire card for the DM4800, as I won't mind as much permanently assigning half of them to physical inputs from the DM.

Thanks again.
 

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