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Funky Fingers 30 January 2011

I am trying to obtain a signal level of -20dBFS to -12dBFS at the input channel on Cubase 5, using the Focusrite Octopre Dynamic MkII mic preamp . The problem that I am having is with the recording of the drum tracks. The signal level that I am getting at the input bus is -1dBFS to -3dBFS even though my gains are set at zero on the mic preamp. While I am not getting any clipping on the preamp or at the input bus I can't attain my desired headroom level of peaks at around -12dBFS unless I adjust the trim knob, at the top of the input channel of Cubase, to a setting of -9dBFS to -12dBFS. I am not sure if this is the proper use of this gain knob on Cubase or if I should be looking for another resolution such as an in line attentuator on the problem mic lines. Thanks in advance for any suggestions.

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Funky Fingers Sun, 01/30/2011 - 19:36

I'm not sure if you understand exactly what I'm saying. My preamp (Octopre) is turned down to the minimum setting, I can't adjust the signal to be any lower from the octopre and therefore my signal level at the input channel of Cubase is -1dBFS to -3dBFS which I assume would be the level that would be printed to the hard drive of the computer if recorded. I want to record and print a signal level that is between -12 and -20dBFS but I can't achieve this with the signal that is coming from the preamp. If I adjust the input gain control (not the input channel fader) on Cubase's input channel it will lower the signal level that is printed. I believe the purpose of this gain is to control the gain for the incoming signal before EQ and effects. I just want to be able to record a signal with plenty of headroom and I'm not certain if this is the best way to achieve this, or if I need to find a way to attenuate the mic signal before it gets to the preamp and by doing that, it would allow me to control the signal with the gain control of the Octopre preamp.

Davedog Sun, 01/30/2011 - 21:48

Pad pad pad. And yes, use the channel gain to pad down the mic signal going into the preamps. Then use the channels attenuaters to makeup the gain to the recorder. If the onboard pad and the input gain doesnt do it then an inline pad of some sorts should do the trick. See John's post for availablity.

RemyRAD Mon, 01/31/2011 - 01:51

I just can't accept the fact that this particular device requires any kind of input pad. Not with the microphones he just mentioned.

This thing also has compressors on each one of these microphone preamps. Nobody's talking about where these things are set and it doesn't sound like it's set to zero. I mean SM57 on snare drum is what this thing was designed to work with. Cue Base should be having no issues with this.

If you're getting peaks without clipping, you don't need any more headroom than that. I don't give a crap about -12 DB FS blah blah that's crap. Take a close zoom in look and see if your waveforms are flattopped? If they're not? Don't fix it! If you want to make use of your maximum number of bits, record hot. If you want to worry about what a stupid book or class has said, recorded at -12 DB FS. So don't be silly because you're being silly. I mean if you need that kind of head room because you are going from a quiet oboe solo to the cannons going off for the 1812 Symphony, that could be a problem. But you're talking rock 'n roll and we are official rock 'n roll engineers. If you're not hearing bad sounds it's not bad. If you're hearing bad sounds, you're doing something wrong and it ain't the equipment. Don't think you want that "more" button pressed on that OctoSuck either. The crappy condenser microphone in the bass drum however IS a problem. It's going to output too much since you have no pad on the microphone. That is unless you do? If you do? Use it. If you don't? Use a $20 directional RadioShack dynamic microphone that sort of looks like a SM58. This will do a far better job and then you can reserve your cheap condenser microphone to use overhead of the drums. But you can't use that in the bass drum it ain't going to work with or without an external pad because the capsule is overloading the transistor within the microphone. So that's a no do, don't go there kind of thing you're doing. And that's not a problem with the preamp and nothing an input pad will solve. That microphone is NOT designed to be used in the application in which you are using it. Sorry kiddo.

It doesn't matter what kind of equipment you use as long as you understand what you are using.
Mx. Remy Ann David

sshack Mon, 01/31/2011 - 07:16

Great point Remy. I actually spend MUCH of my time these days looking at my recorded wave form, more than just looking at the track levels alone. Invariably though, they do go hand in hand...at least in my experiences. After much trial and error I typically leave my track faders in Logic at "0" and try to get my signal somewhere between -16 and -6 depending on what I'm recording and how dynamic it may be. The checker for me though is always looking at the wave form.

As it's been discussed many times in the past, I think most of us feel like we need to have these big huge wave forms filling up all that space on our tracks, which may be acceptable for say a single acoustic guitar recorded with just vocals, but as soon as you start trying to stack up a few tracks like a normal rock song (vox, drums, guitars, bass, etc.), you're immediately in the mud. Especially if you intend to use compression at any point be it on an individual track or on the bus.

Big K Mon, 01/31/2011 - 08:16

You should re-think the way of recording with DAWs.
Nothing is easier than increasing or decreasing recorded levels.
Waveform is not interesting, either, maybe as fast visual reference and for me not even that.
Meters are what counts, because the fast overs are hardly seen and easily overlooked.
It does not matter were you set your personal max level point, as long as you can record safely.
In DAW world you have lots of stages to deal with levels and for reasonable handling of Compressors and the
like you want a decent level, anyway.
Why would I be in trouble with a few dozen tracks of rockmusic at -4 to -0 dB?
Logic seems a strange software...lol...

sshack Mon, 01/31/2011 - 09:45

Well, there's always more than one way to skin a cat, right? Truth be known, Logic is fine, I'm the strange one.

The art of gain staging is something that I would consider a life long school and since I use all outboard gear (not many plugins) there's always a knob to turn, tweak or adjust in the signal path. That's not to imply that I never touch my track faders in Logic, I just tend to keep them at '0' for tracking and adjust as needed as I progress through the mix.
When I first started recording, many of my friends that were already into DAWs told me to record as hot as possible without clipping...so I took their advice. As time went on and I learned/experimented more, I just felt like I didn't really care for that approach as I didn't really like the sound; I started to back way off on my levels. As a result, I'm not so concerned that my mixes are loud, or louder than the next guy, I'm just after clarity and depth as it pertains to the song I'm recording.

Again, many ways to skin a cat...way too many variables to nail down a "right" way.

So maybe my cat has an inverted Mohawk?

mdb Mon, 01/31/2011 - 14:19

What does the OctoPre run through? I'm not familiar with it so I don't know if it acts as an interface or not. Whether you have a separate interface doesn't really matter (unless you do and it's gain is set all wrong), is the gain stage on your software mixer also set appropriately before it gets to Cubase? Like others have said, there's a lot of places that the gain could be off within the signal path to the DAW. The problem definitely isn't the DAW or the SM57. The SM57 is a dynamic mic and requires hi-gain to work well.

Big K Mon, 01/31/2011 - 15:31

Hello sshack
quite true ..many ways.. But, if a recording at -1 dB sound different to you ( apart from the volume) than one at - 10 dB there is something wrong with the converter.
If you do a phase reverse test like record at max -10 dB and record at max -1 dB, reverse phase on one channel and lower the higher level to fit the lower exactly, it should
cancel out to bout nil. If it doesn't, you have a problem with the pre-amps. I say about, because with some pres there is a slight difference in sound when driven, but this
should not be a significant one.

Of course, with 24 bit or 32 bit floating you can record at -20 dB if you like, and you can still bring it up to -1 dB without compromising the sound.
But for the settings of several FX sake, a reasonable level to work within adequat setting is desireable. I am sure , you know that and you do a fine
work with the workflow you made your own. But as you said, many ways ...

RemyRAD Mon, 01/31/2011 - 18:39

I understand what you're also saying BigK. But also consider this. Some of us, have come a long way since the days of Scott 111 & Ampex-300, to my current hard disk 24 track recorder. You really get different sounds on each one of these recording devices differently, at different levels. When we were all originally restricted to 16 bit 44.1 kHz, you really wanted to have your maximum resolution. Maximum resolution was only possible at the highest recording levels possible. Of course, dynamic range dictates that everything cannot be kept at a steady level of 0 DB FS. When you recorded at lower levels, you were actually recording with lower resolution since you cannot make use of all of the available bits. Of course even when later level compensated, there is an obvious difference. The slightly lower-level recordings that were lower in resolution may actually be the auditory analogy of the way to make digital sound more like analog. Higher resolution also means greater detail in what's not right. So who wants to really exaggerate their inadequacies? Unless there is a reason to do it such as a smoother sound. This is all very subjective that's what's exciting about audio. Sure, you can tell your equipment to choose 24 or 32 bit, floating or sinking and believe that you'll have the recorded resolution to compensate for it in the end. I don't necessarily agree with that either as the amplifiers themselves all go nonlinear in different ways, different levels, different times. So it will sound different again and with each variable. There are some places that we all like to be in and that works best for us. My grandfather once told me, insisted that, you don't put the ketchup on your hamburger. No! You put the ketchup on your plate and you "smoosh" your hamburger edgewise through the ketchup! I never did that. I still don't do that. And he's dead now so I guess I win?

You have to know what you want. Then you have to know how to get there.

Turn left at the next decibel. Remember when you see a fork in the road, add +2 DB at 10K.
Mx. Remy Ann David

Big K Tue, 02/01/2011 - 06:36

Hi Remy...
After years of recording through analog consoles to MT Tape machines and to digital Tape of a 3324S, I know exactly what you mean. I also wrote, and agree with you, that mic pres can have small differences in sound when driven at different levels.
Buuut, ... would you want to use an amp that makes his own noticeably different sound between -10 and -1? I'd kick its butt..lol... since the normal dynamic during recording is much greater and I surely want no difference in sound there.

:-) I put my ketchup on the burger, too.

---------------------------------------------
For long, I have stopped using any dynamic gear ( only if I want that particular hardware sound ) in the signal path to the DAW and keep a safe "headroom", myself, but I am no fan of selfimposed rules. For what I see and hear everyday, once the digits are in the box, a good DAW audio engine can adjust almost any recording level, even to some extremes, without changing the sound with 24/32. It is just Math, which is what computers love to do and are pretty capable of.
The important bits are surely the source, mic, cable ( hello audiokid..lol :) and pre-amp.
Anyway, I am not preaching anything and there are not many inevitable rules to obey, either. Audio is amazing and the workflows to get to reasonable results are many.
Peace...

Still, why does he have too high levels in Cubase... with a SM57?
Any news ?

Big K Tue, 02/01/2011 - 13:20

Hmmm, I can't comment on that... I am sitting in a glas house..

I had such a bloody stupid annoying ( for unlawful carnal knowledge ...ing ) situation just today.
And it was just a stupid button... I rather call the button stupid then myself....old habit.
;-)
Am I getting old, or am I still not old enough, yet???
Sigh....

Funky Fingers Wed, 02/02/2011 - 13:42

Hi everybody! I have been away for a couple of days and yes my issue is still unresolved but I do have some things to try based on all the replies from everyone. Remy you did bring up some valid points, I agree that your max recording peaks can be hot as long as they are not clipping, when I first got into this home recording I used to record as hot as possible w/o clipping. There is a great debate about the validity of recording with more head room and there are many voices on each side of the argument. I'm not saying either way is right or wrong, perhaps it is really comes down to personal preference and what each person believes their ears to be hearing, however that's not really the issue here. I should be able to attain whatever signal level I am after and that has been my issue. The apex 125 mic that I mentioned above is in fact the recommended microphone for kick drum in the Apex Studio drum mic starter package. I agree that it is a very crappy mic and I have been working on upgrading one by one. I do however disagree that a $20. radio shack mic would be better, as good.......maybe? I will confess that the compression settings on the Octopre are being used with the "more" button depressed and perhaps this is contributing to my problem but I'm not convinced that I could still achieve the head room I'm looking for ( even though you think I'm being silly.....but maybe I am ).

TerrorRun, you mentioned to use the pads on the Octopre. The Octopre Dynamic doesn't have the pad button as does the Octopre MkII. I am running this through Focusrite LS56 using the ADAT connections and I'm recording at 44.1 kHz.

TheJackAttack Wed, 02/02/2011 - 14:14

Ok here is the thing about compressors, a stand alone unit generally has many different adjustment knobs. There is an input gain boost, the knee, the ratio, the make up gain. I was aware your Octopre had built in compressors but the key software/hardware knob that is too hot is the make up gain. If you do not have control over the makeup gain then I wouldn't use the in built Octopre compressor.

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