Can somebody help me please?! with my RODE NT2- A

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tupac0306

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I am sorry. I am really a beginner in recording. to make it short .I pair up my NT2 A with Mbox 3 mini to my macpro. I use logic studio. In order to get my mic loud enough I have to turn the gain at least to half way.but the mic sounds really tinny and if I speak a lil louder the mic produces some noisy sound (I dun know how to describe it, it's when I sing in a louder volume, the sound is not clear enough..it's noisy) Can someone tell me why is this?.If I turn the gain down. my voice will be completed covered by my song instrument (I can also turn down the instrument volume, bt that will make the whole song really really quiet) What should I do?

Do I have to get a mic preamp? (I believe Mbox 3 hv it already inside). If i have to, what's very suitable to NT2-A
I am recording in a quite big room. Should I use a reflection filter to filter out the unnecessary echo or noise?

I will try it out with my apogee duet 2 which will arrive this saturday. hope it can get better. Bt I honestly think I didnt do it right.

Sorry If my question is silly to you..I just don't know wt to do:o

please help me. thank you so much!

J
 
if you run an insstrument such as a guitar through the mbox how does it sounf?
 
Your instruments were tracked at too high of a level. Knock the faders down on the individual tracks so they peak at about -18dB to -12dB.

Track your vocal to peak at about the same.

The Mini specs say the mic pres have 54dB of gain w/out the pad. Turning the input halfway up is overloading the input on the Mini.

I have an NT2 and use preamps with about 60dB of gain and only need to turn the input volume up to around 9 o'clock---i.e. about 1/4 of the available gain for vocals.
 
It would be a little hard to make calls on levels without any samples. If you're mixing tracks, you want them a bit less hot before mixing or the additive effect will clip. Which you can do while tracking or in post. There is noise associated with most recording gear. But that gear looks to be mostly okay in that regard. The mbox being the question mark IMO. The spec sheet seems to suggest that there's a significant difference between the PRO version and the other versions. The base model seems to not handle a hot input signal according to specs, which might be what you're experiencing. You might try using the pad switch on the mic. Or some other form of attenuation and see if that helps.

http://www.avid.com/static/resource...Avid_Mbox_3rdGen-family_spec-comparison-1.pdf

How bad does a company have to be to have to URL mangle manually to actually get the correct documentation link from their webpage. i.e. That's not the link actually on this page: Avid | Mbox Mini

RØDE Microphones - NT2-A
http://www.rode.com.au/downloads/nt2-a/nt2-a_datasheet.pdf

Plenty of SPL tolerance and low noise at least in specs for the NT2-a. So it's probably not the mic. You might still need a pop filter to prevent certain noises. And a shockmount for more of the same.
 
c7sus said:
Your instruments were tracked at too high of a level. Knock the faders down on the individual tracks so they peak at about -18dB to -12dB.

Track your vocal to peak at about the same.

The Mini specs say the mic pres have 54dB of gain w/out the pad. Turning the input halfway up is overloading the input on the Mini.

I have an NT2 and use preamps with about 60dB of gain and only need to turn the input volume up to around 9 o'clock---i.e. about 1/4 of the available gain for vocals.
Click to expand...
If i knock everything down. How can I make the whole thing sound louder after I record. If I use -18db to -20 db. They actually sound will less than a half of the volume of a ordinary track which I won't be to play it loud using any music player.
 
Just out of curiosity, doesn't the mbox have a (+48v) phantom power switch....if it does, have you made sure its on. Sorry for stating the obvious, but you don't know if you don't ask.
 
Richie2708 said:
Just out of curiosity, doesn't the mbox have a (+48v) phantom power switch....if it does, have you made sure its on. Sorry for stating the obvious, but you don't know if you don't ask.
Click to expand...

Yeah It's on. I turned on the 48V switch dun even use the gain.The microphone is at the position of 0 dB. If I just record a single vocal track only. It should give me volume like at least about -5 dB right? It turned out give me like - 20 dB recording volume. So there must be something wrong either with the mic or the Mbox right?
 
Process of elimination will help. Ask a friend if you can try your mic on their system, then try one of their mics on yours.
 
Proper Audio Recording Levels | Rants, Articles | MASSIVE Mastering

There's nothing wrong with the mic or the preamp. What's wrong is your understanding of audio levels and the way you are using the gear.

So - You have a microphone and a preamp going into a converter or sound card. Those converters are calibrated at LINE LEVEL. In most cases, over the last several years, most I've seen are calibrated to -18dBFS = line level (or 0dBVU). In other words, if you run a steady signal (a sustained note on a keyboard for instance) through a preamp and turn up the preamp gain until the VU meter reads 0dBVU, at the converter, it will read -18dBFS (or -18dBFS(RMS) -- full scale, but measured over time).
 
Thank you for the reply. So the VU meter in my logic studio is in the unit of dBVU I suppose right? I did recorded a sustained note (with my mic at 0dB position, with no gain from the Mbox). The VU meter in my logic studio actually says about - 20 dB. My Mbox cannot even read any dBFS data. Or did you just mean that my logic studio will just read my interface's signal which mean - 20dB is roughly about 0 dBVU? Even that's the case. My microphone still sounds really really quiet at this level line. U can hardly hear your recording unless you record really close to the mic.
 
Logic is calibrated to dBFS.

So when the MBox is outputting 0dBV Logic should be seeing about -18dBFS.

How about posting a link to a clip on soundcloud or something? My take on all this is you tracked a bunch of soft synths and drums and whatever and they are all too hot to begin with. When you try to match your vocal to the already too hot instrument tracks you are getting distortion from overloading the pres on the interface and maybe even digital clipping from trying to track the vocal as hot as possible.

The final volume level of your finished track is determined at the time of final mixdown and mastering, not during tracking of the individual tracks. Read the blog again from Massive Mastering. If you tracked 10 tracks at -3dBFS and then played them back with the software faders at unity gain your master stereo output would show digital clipping--i.e. going over 0dBFS. Why? When you understand the "why" of that you will begin to understand headroom.

In fact, you can quickly see this for yourself. Take your instrument tracks and put all the faders at unity gain-- 0dB in Logic. Now just make a quick stereo mix of all those tracks with the master set at unity gain. It's gonna clip or be damned close.
 
Yeah I see what you mean. Sorry about being stupid.Thank you. So is that mean If I track all my tracks properly with same appropriate line levels. Even they sound really quiet in logic studio. The finished track will play louder?
 
-18dBFS to -12dBFS. That's dry tracking with no compression.
 
tupac0306 said:
Yeah I see what you mean. Sorry about being stupid.Thank you. So is that mean If I track all my tracks properly with same appropriate line levels. Even they sound really quiet in logic studio. The finished track will play louder?
Click to expand...

It's not being stupid, it's just part of the learning curve with this stuff. It doesn't help that if you are using softsynths the manufacterers use samples that are processed to a certain degree. I always have to turn down samples from both Dimension Pro and EZDrummer because they will clip or distort otherwise.

If your stuff is sounding really quiet you need to check the levels of the output faders in the Logic console. I run the amp to my speakers a bit hot so I seldom have any hardware or software faders set all the way to unity gain. Usually individual faders are between
-6dBv to -10dBv.
 
Hi there tupac0306.
I have an NT2A, bought it a few months ago. At first i thought it was me doing something wrong too. Though I have been recording for 40 years. I use Apogee duet. Same problem background noise. It didn't appear at first but after a few hours of use. It has been back for 2 services & now the noise is back. Other mics work on same set up so I'm sure it is the mic.
Did you get your problem sorted out, if so how. Interested to hear your reply.
Thanks
 
HI there,

I have setup a home studio. I have a MacBook Pro connected to Avid Mbox Mini Soundcard, two speakers and a Rode NT2 condenser mic.

For some reason, the voice or acoustic recording level is very low...I can still record but the sound level is very low and if I increase the input volume then there is a terrible echo...I am sure there is something wrong.

These are the solutions I have already tried:

- System preferences, I put the input as Avid Mini...and activate +48 button to power the mic.

- Garage band, I move track volume to track pan, Monitor ON, 24bits...and the Recording Level is gray-out since I am using an external sound card (I assumed because of what I read).

- I have also moved from Garage Band to Ableton but I have the same issue...I can record only at max gain and I also here my self only at max gain...

Any idea on how I can solve this issue, I am sure there is something wrong, the mic and soundcard can record this bad.

Thanks in advance. Mariano
Mac Pro, Avid mbox mini NT2 Rode
 
Turn off your monitors and monitor through headphones.

The NT2 is a pretty sensitive mic. My main preamp has about +58dB of gain and I only need to turn up to about 9 o'clock to get a good solid signal.

You could *possibly* have a problem with the phantom power on the Avid or a loose wire in the Rode.

Are you using an XLR male > XLR female cable? Could be a bad solder joint in the cable too.

But first I would turn off your speakers and monitor through headphones. That should solve the "echo" problem, which is really a feedback loop created by having the mic on while listening through speakers. Also, set the asio driver to the fastest setting. That will reduce latency which can also produce an echo-like effect because of the recorded signal lagging the input signal. You want to figure out how to monitor the direct signal and not the signal after it goes round-trip through the Avid. Adjust the mix knob on the Avid so you are listening to the direct sound and not the sound written to disk. Dig?
 
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