Line in from Mic pre to DM3200

kei_siuip

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I've recently resettle my studio with DM3200 and PT 10. I patch my NEVE mic preamp output to TubeTech CL 1B compressor. And the output of TubeTech to line in of DM3200. The signal is successfully routed to ProTools through IF-FW card. However, I still have to adjust the gain knob to 10 o'clock of the channel to get the signal loud enough. Did I do something wrong with my setup? Because I think the gain knob shouldn't be adjusted if there is 'Line In'. Also, there are only 24 I/O shown up in ProTools rather than 32, why??
 
You'll need to be more specific. How exactly are you patching the Neve and the TubeTech compressor? Describe specifically the wiring - mono, stereo, etc - between those two devices, and the OUTPUT from whichever device goes into the DM3200.

You're right; there should be NO reason to increase the DM's pre amp pot unless there's a signal deficiency somewhere along the line.

CaptDan
 
You'll need to be more specific. How exactly are you patching the Neve and the TubeTech compressor? Describe specifically the wiring - mono, stereo, etc - between those two devices, and the OUTPUT from whichever device goes into the DM3200.

You're right; there should be NO reason to increase the DM's pre amp pot unless there's a signal deficiency somewhere along the line.

CaptDan

I plug the 'Line out 1' of NEVE, to the single input of TubeTech. And the only output of TubeTech, to the 'line in' of channel 1 of DM3200.The whole thing is mono.
 
How are you setting the TubeTech? Are you selecting an extreme compression (say - 10:1, threshold: -20db, attack <>250ms, release 500ms, with NO OUTGAIN compensation?). If so, try boosting the TubeTech's gain a few DB; your goal is to achieve a peak signal in the DAW around <>-6db maximum. If the sound is too low, there are several ways to boost it to the desired level.

If that doesn't help, the next thing to do is BYPASS the compressor by testing the Neve directly into the DM. If you achieve the right signal level without the compressor inserted in the chain, then there's an issue with cabling, physical routing, or perhaps, a malfunction with the TubeTech itself.

CaptDan
 
How are you setting the TubeTech? Are you selecting an extreme compression (say - 10:1, threshold: -20db, attack <>250ms, release 500ms, with NO OUTGAIN compensation?). If so, try boosting the TubeTech's gain a few DB; your goal is to achieve a peak signal in the DAW around <>-6db maximum. If the sound is too low, there are several ways to boost it to the desired level.

If that doesn't help, the next thing to do is BYPASS the compressor by testing the Neve directly into the DM. If you achieve the right signal level without the compressor inserted in the chain, then there's an issue with cabling, physical routing, or perhaps, a malfunction with the TubeTech itself.

CaptDan

THX.
The Tubetech setting is moderate and normal, threshold around -10, ratio is 3:1, attack and release is moderate too. The output gain is ZERO.
Actually, before using DM3200, I patched the Tubetech out directly to the input of Digi003 rack. Everything is normal.
I'll try again tomorrow to see what happen.
 
Is there a +4/-10 setting on any of the equipment outputs? The DM line-in is expecting a +4 signal. If the TubeTech is set to output at -10, that may be the problem.
 
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Is there a +4/-10 setting on any of the equipment outputs? The DM line-in is expecting a +4 signal. If the TubeTech is set to output at -10, that may be the problem.

No output setting on NEVE and TubeTech
 
How about the digital input trim settings on your DM? are they also set to 0db?
 
Have a look at page 71 in the 3200's manual. The digital trim is a way to adjust the digitalized signal level, just after it has gone through the A/D converter. It has nothing to do with anything before the converter. A quote from the manual:

"Digital trim Allows the module’s input to be
adjusted. This is post-phase and pre-gate. It is
entirely independent of the analog TRIM controls on
the mic/line inputs.
The value of this digital trim can be adjusted from a
cut of –50 dB to a boost of +10 dB in 0.5 dB steps.
Also see “Phase/trim/delay” on page 71 below."

I believe that the default level is 0db. If your digital trim is set too low, you may be trying to make up the gain via the analog trim knob. It's just a thought....a good thing to check if you haven't yet, but the problem is probably solved elsewhere.
 
In PT10 go to the I/O page and click default - that should make the additional I/O show up.

If you are patching your Neve/TT into a M/L in on the DM I'd expect that you'd need to set the input gain on the DM (ideally) to unity or a little higher than the level being sent from the Neve/TT. Are you using the TRS inputs? Did you try using one of the auxilliary returns instead of a M/L in?
 
Have a look at page 71 in the 3200's manual. The digital trim is a way to adjust the digitalized signal level, just after it has gone through the A/D converter. It has nothing to do with anything before the converter. A quote from the manual:

"Digital trim Allows the module’s input to be
adjusted. This is post-phase and pre-gate. It is
entirely independent of the analog TRIM controls on
the mic/line inputs.
The value of this digital trim can be adjusted from a
cut of –50 dB to a boost of +10 dB in 0.5 dB steps.
Also see “Phase/trim/delay” on page 71 below."

I believe that the default level is 0db. If your digital trim is set too low, you may be trying to make up the gain via the analog trim knob. It's just a thought....a good thing to check if you haven't yet, but the problem is probably solved elsewhere.
The Digital Setting is 0db
 
In PT10 go to the I/O page and click default - that should make the additional I/O show up.

If you are patching your Neve/TT into a M/L in on the DM I'd expect that you'd need to set the input gain on the DM (ideally) to unity or a little higher than the level being sent from the Neve/TT. Are you using the TRS inputs? Did you try using one of the auxilliary returns instead of a M/L in?

I know how to click default I/O in PT. It's still 24 I/O only.
I've used TRS input.
 
It sounds like you have the original firewire card - the IF-FW/DM. The IF-FW/DMMKII is the one that does 32 tracks. This is the exact text on the DM3200 page on the Tascam site:

"The DM-3200 provides a new way to record with DAW recording software. With the addition of an optional expansion card, the IF-FW/DM FireWire interface, the DM-3200 becomes a 24-channel computer audio interface, making it a complete studio solution for professional DAW recording."

You need the IF-FW/DMMKII.
 
Last edited:
It sounds like you have the original firewire card - the IF-FW/DM. The IF-FW/DMMKII is the one that does 32 tracks. This is the eaxct text on the DM3200 page on the Tascam site:

"The DM-3200 provides a new way to record with DAW recording software. With the addition of an optional expansion card, the IF-FW/DM FireWire interface, the DM-3200 becomes a 24-channel computer audio interface, making it a complete studio solution for professional DAW recording."

You need the IF-FW/DMMKII.

Oh, Got It!
 
We're now testing the talkback and foldback from the mic in to the dubbing artist. I find that the internal talkback mic get heavy noise if I turn the talkback level to normal, should I change that mic?
Also, I set the foldback from mic in to dubbing artists through AUX1-2. I find out that it sounds like Echo/Delay. I route the input signal of mic to BUS 1, then to the ProTools. Should I route the holdback from BUS 1 to AUX1-2?
 
The DM's talk back mic is - shall we say - challenged. :X3: It tends to sound better if it's used in an 'inconvenient way' - meaning, with the volume decreased and spoken into with a semi-whisper, 4 inches from the mic. Not particularly wonderful. If you have a simple, dynamic mic (SM57) available, you can mount it on the console, and route it into the Stereo Bus, using the selected channel module's Mute button as the 'talk back key. The DM's 37-40 'utility faders' are ideal for that.

Echo/Delay; sounds like you're monitoring the ProTools track instead of a split input on the DM. If you do this, you'll get noticeable latency. This has been discussed a lot; do a search here for 'Input Bypass.' Even if you're not using Input Bypass, several of those messages include instructions for setting up a latency-free monitoring channel on your DM.

CaptDan
 
Ever since I bought my 3200, I was unimpressed with both the location and sound quality of the built-in TB mic. Right from the start, I went to a local music store and bought a sure PG58, their cheaper version of a 58, and have it plugged into M/L 16. I typically have it mounted on a small desktop mic stand, and keep it placed on my monitor shelf when I am not using it. When I do use it, I pull it off the shelf and keep it near me by the mixer. To use it I have/use a foot pedal plugged into the back of the DM that turns the mic on or off. Very convenient and easy, with much better sound quality than the built in one. It picks up voices behind me in the CR as well, when one of the listening musicians wants to talk with the one tracking. It does use up one mic input, but that has not been an issue.
 
The DM's talk back mic is - shall we say - challenged. :X3: It tends to sound better if it's used in an 'inconvenient way' - meaning, with the volume decreased and spoken into with a semi-whisper, 4 inches from the mic. Not particularly wonderful. If you have a simple, dynamic mic (SM57) available, you can mount it on the console, and route it into the Stereo Bus, using the selected channel module's Mute button as the 'talk back key. The DM's 37-40 'utility faders' are ideal for that.

Echo/Delay; sounds like you're monitoring the ProTools track instead of a split input on the DM. If you do this, you'll get noticeable latency. This has been discussed a lot; do a search here for 'Input Bypass.' Even if you're not using Input Bypass, several of those messages include instructions for setting up a latency-free monitoring channel on your DM.

CaptDan

Well, we can't simply use a dynamic mic fortalkback, because some clients sit on the sofa behind me and give instruction. The distance may be 1-2 feet. When I switch on the talkback, it sounds like playing the -20 pink noise at the same time.
The foldback problem have been solved, thanks
 
some clients sit on the sofa behind me and give instruction

Excuse my off-topic swerve, but that reminded me of a funny story:

There was a famous American TV commercial director - one of the highest paid in the industry. He was so powerful that - unlike most other directors - he'd insist that nobody from the ad agency could be on his set while he was working. That was unprecedented, because ad agencies are very 'hands-on.' They like to send an entire squad to a production, just to 'keep tabs on things.' (They have an expense account, after all. :))

So one day he's rehearsing a scene, when suddenly an ad agency 'A.E.' (account executive) pipes in with his opinion. So 'Rick' picks up his BULLHORN, and shouts:

"LISTEN UP!! I NEED JUST ONE MONKEY FROM THE AGENCY!! NOW YOU F****S DECIDE AMONG YOU WHO THAT MONKEY'S GONNA BE. THEN SEND HIM BACK.....GAGGED AND BOUND. OTHERWISE, ALL YOU $#21- GET THE *&^%$%#$# OFF MY SET!!!" :eek:

Now - I'm not recommending that you do the same thing with your studio clients. Just hoping to send a laugh your way. ;)

CaptDan
 

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