Haha... "Overly simple" never describes gain staging. I see it all the time, "engineers" with more experience than myself don't know what they're doing with this.

But a good rule of thumb I've read, heard from, and experienced myself is to record at much lower levels than you would inherently think of. Generally I limit my peaks to -12db. A professor of mine swears by being able to peak around -30db. The lower you record, the more headroom you have. In the digital age, our noise floor is extremely lower than that of analog. We don't have to worry about tape hiss, and all that jazz. What we do have to worry about is what else is getting in our signal, like mic self noise, anything else going on in the room, noise from your LA-610 (yes it has noise, all amps do), etc. But if you've got a good signal chain and you've controlled your rooms noise floor a bit, you shouldn't have any trouble recording at really low volumes. And then you'll be able to put all the EQ, compression, limiting, delays, reverbs, autotunes (joke! kinda...) that you want!

There is one downside to this however. Most gear, and even DAW's don't really balance things out for you in the lower volume areas. Meaning you've got exponential control on faders, notice the large distance between 0db to -20 db, and the small distance between all that lower stuff. So it can sometimes be difficult to be really accurate with your monitoring.

But seriously, "no blood on the tracks" is a good philosophy to live and die by in recording. The louder you record, the more limited your dynamic range will be. I personally still stick to -12, I think it provides plenty of headroom and a good signal to noise ratio.

im a big fan of this plug in.
i use it to set the recording levels on my tracks.
i also like my peaks around -12....average -18

Download Inspector Free by Roger Nichols Digital

one of the things that I've done to guard against this is to automatically pull the faders down for ALL of my instrument tracks in Logic from "0.0" to "-0.6".

I hate to tell you this but there is no point to bringing the faders down in Logic. All of your gain staging has to be done before it even get's into the box.

Are you still using the Duet? If it is possible to control the input levels to the interface that would be where to turn it down. As it is, I think that you'll find once you turn your faders back up to 0 it should be fine. If it wasn't clipping with the faders down, it shouldn't clip with the faders up. All the faders do is control the output level of each channel/track.

This is something I toss around in my head from time to time, as well.
Not adjusting the DAW input faders, but gain staging related to DAWs in general.

I have pretty much the same approach as sshack.
I typically go for peaks between -16 and -12dBfs (sometimes reaching above -10dBfs), while shooting for an RMS level around -22dBfs. (Also typically ends up a little higher, of course.)
I'll also hit the snare, kick, or bass (2 of the 3, depending on source) w/ my dbx comps to tame the peaks a little. Other than that, I go au naturel.

Disclaimer: I know there are no "textbook" levels at which to track.
My main concern is the gain staging related to digital.
I use Cubase SX3, and have the option of mixing in 24-bit or 32-bit float.

So here's my question, and I think sshack's for the most part:

Does your level coming into the DAW (they are usually similar to the final analog level, but each interface/DAW will show different levels)
vary if you're working w/ 24-bit vs. 16-bit vs. analog (tape, etc).
Shouldn't the added bits give more headroom, and allow one to track a little hotter (so long as there's no clipping)?

Can of worms, I know.
I'd rather get the experiences of the people here, that I KNOW use the stuff and practice it daily, rather than that of the people trying to get me to buy it.

I have some thoughts - I hope these are helpful. I don't tend to track at different levels between 16 and 24-bit but in theory I think you could track at a lower level in 24-bit and still have the same or better dynamic range that you would in 16-bit. So in otherwords, I don't think it's as much about headroom as it is the range of 0 to loud. 24-bit allows you the freedom to sample audio at a quieter volume, allowing you to more accurately record the original dynamics of the performance.

With that thinking in mind, I will typically set my levels fairly conservatively going into my pres and to the converters (particularly for percussive instruments like piano). Once the signal is in the digital domain I manage the signal at each processing stage to make sure it's not clipping anywhere within the sofware.

I'm not sure if this is helping.. Here's another way to think about it. It's not as much about getting a "hot" signal recorded. That was the way of thinking with analog as analog tape gives you some margin of error if you start to distort (headroom). With digital, it's about getting a "clean" signal recorded. So gainstaging before conversion is important to keep noise floors and distortion levels low. If you get clipping at the conversion stage with digital, it is unforgiving,

Once you're in the digital domain, think of 0dBFS as your absolute limit (I actually prefer -0.5dBFS and only for the loudest peaks). So if your signal was recorded at -20dBFS, you have some room if you EQ, maybe compress to increase and even out the volume of the track. You'll want to constantly monitor/manage to make sure that any of the processing being done in the software isn't creating "overs" at any stage though.

The way to get a "hot" sounding mix in the digital domain has more to do with managing the overall spectral content and dynamics of the mix as opposed to gainstaging before A/D conversion or recording levels during tacking.

Cheers :)!

There can be advantages to recording at 24-bit but some of these advantages cannot really be produced. 16-bit nets you 96 DB of working level. 24-bit provides you with 140 DB of working level. But here is the catch in the get along. THE ELECTRONICS THEMSELVES GENERALLY CANNOT PRODUCE MORE THAN 110 DB OF USABLE RANGE. So what about those other 30 DB? In a sense it's like vaporware. This is what our software is going to do when we can figure it out, maybe, etc. It allows for some better processing at levels that would naturally be below the noise floor of 96 DB. But if it's still that low, you won't hear it in the mix, not until you raise the level and then still have to deal with whatever extra low-level noise you will introduce. So 24-bit can actually provide for false confidence. I frequently record hot digitally and I am not grossly upset when short duration transients may clip. This can actually be advantageous to certain kinds of recorded material. Most monitor systems provide between 5% to 10% speaker distortion on average. So you are not going to hear a 1 or 2% of harmonic distortion when your monitors will only present you with a minimum of 10 to 20 DB of harmonic distortion.

Not all distortion is bad or inappropriate
Mx. Remy Ann David

hueseph, post: 347414 wrote: I hate to tell you this but there is no point to bringing the faders down in Logic. All of your gain staging has to be done before it even get's into the box.

Are you still using the Duet? If it is possible to control the input levels to the interface that would be where to turn it down. As it is, I think that you'll find once you turn your faders back up to 0 it should be fine. If it wasn't clipping with the faders down, it shouldn't clip with the faders up. All the faders do is control the output level of each channel/track.

Don't hate, I'm here for the learning. :-)

I actually have an RME FF800 now, which is capable of doing WAY more than I need or know how to use. In fact, I would venture to say that I'm probably not even using it optimally yet as so much of it is over my head and I'm hard pressed to find real world examples or tutorials that I can apply to my scenarios/situations.
And quite honestly, I've never even given the 'bit' aspect of this conversation much thought. I'll admit, I don't fully know the difference either. No pride here. I'd love to be educated in these areas of recording and enginnering, but I'm not and unfortunately I don't learn as well by just reading. At the end of the day, the majority of my learning has come from experimenting....just turning knobs, moving mics, trying stuff out.

It's been so long since I've left my faders at "0", I'll just have to go back and try it. Again, it seems in so doing that I would have to scale back on driving my preamps, particularly if I want my peaks around -12.

I appreciate the conversation and input from everyone though.

If you don't want to scale the preamps back then buy yourself a set of XLR pads. They come in 10dB, 20dB and 30dB varieties. Then you can keep your preamps cranked and still have the levels low. But in fact Hueseph is correct. The digital gain controls in a DAW don't affect the signal coming in at all.

As to learning any interface, the best learning proceedure is to have it all set up. Then read for a bit followed by experimenting with that aspect of what you just read. Lather, rinse, repeat.

No worries...I'll just go back and experiment some more. The thing that probably moved me to post this is that I just bought a drum kit and am now venturing into the realm of trying to get a good recorded sound. I'm in the middle of trying just about every mic and pre combo that I have, including padding said mics (when applicable) and/or preamps. It's certainly a whole different world from my norm of recording guitars and vox, but I'm loving it.

Hopefully I can post a sound clip soon for critique.

The faders you are pulling back: are they in your DAW or are they in the Fireface's mixer software. You should be able to pull those back. That would be the right place to do so.

hueseph, post: 347541 wrote: The faders you are pulling back: are they in your DAW or are they in the Fireface's mixer software. You should be able to pull those back. That would be the right place to do so.

Yeah, see...I'm not sure about the faders in Total Mix (FF interface) either. The faders that I'm referring to pulling down are in Logic, so yes, the DAW. I've tried messing around with the levels for the FF800 and honestly, it only confused me. Are you familiar with the RME?

This is where I feel like I'm doing something wrong. When I record, I get good tracks, but I wonder if I knew what the hell I was doing with the levels, FF800 included, that I could get an even better recording.

I'm all ears......

Most of the mixer software for the interfaces are based on the same concept. Turning down the fader in RME's Total Mix should lower the level going into your DAW. That would be the right place to adjust the input level. You still have to watch for clipping but this will allow you to push your preamps for sure. Thanks for bringing this up by the way. It's something that I've never tried with my DAW.

The Totalmix controls CAN affect the levels depending upon which controls you are adjusting. To really help out your understanding go to the matrix and click a few things. Play back or record through those channels whichever and then go to the mixer view and watch what is blinking. Pull those faders up and down and check it out. You can also adjust pan settings too. I have mixed down sessions via the FF800 rather than a DAW for instance and I know Cucco has utilized the Totalmix to provide three or four different headphone/2 track mixes before. The trick of course is to experiment extensively and write questions and notes to answer those questions prior to using a new configuration at a session or live gig.

I recorded some scratch tracks yesterday to experiment for that very reason and really didn't see that adjusting the faders in Total Mix made any difference to the levels in Logic. I went back and watched some RME tutorials as well and I think Total Mix is really more applicable to providing mixes for overdubs, etc. There are a few more things that I want to try tonight though just to flush out some ideas.
Another aspect is how the FF800 handles I/Os from my outboard verb and compressor (which, technically I won't get until Wed)...but I have messed around with the reverb (LXP-1). It really just boils down to me getting my head around how things get routed into and from the FF, into and from Logic, back out and so on.
Once I 'get it', I'll have it.

Maybe Cucco will chime in at some point and set me straight.

You guys are helping me think through it though, which I appreciate.

When you're recording through the FF and if you have selected a mono track, then obviously there is no pan setting etc. When looking at the Totalmix mixer, the top row is the gozintas-what's coming into the FF. The bottom row is the gozoutas-what the FF is sending out to the hardware/speakers/headphones/etc. The middle row are returns from your DAW-these are useful for sending individual channels back out for submixes or headphone mixes or out into a summing box for analog mixdown. The first thing I would do is practicing routing your recorded tracks back into the FF as individual sticks and play with them that way, sending them to various outputs or singly to the headphones. Play with the pan and faders etc.

i had a look at the FF800 manual and it's a lot like the my onyx 1200f.
total mix is only for setting up sub groups and heaphone feeds as far as i can see.
levels are set on the hardware box through the trim pots.
ajusting the sliders in logic only determine the level of the master bus output.

the inspector plug i sugested works great, just insert it on the track your recording in logic and bring the hardware trim pot up for that channel till you have the level you want on the inspector plug.
that will be your true imput level for that track.
i have a template with the inspector plug on every track. the nice thing is you can use it to set your imput levels and then on playback(when you add compression and e.q. and such) you can see how much gain your adding to that track so you don't clip .
then when i mix down, i put it on the master bus and i can see what all my tracks and master bus plugs are adding up to. i don't like to go past - 3 to -5 there, unless im doing a final mix for myself and using waves L2 or something like that, i'll go -1.

just a sugestion.

boxcar, post: 347593 wrote: levels are set on the hardware box through the trim pots.

There's some handy info.

levels are set on the hardware box through the trim pots and the plug in.lol
i know that in nuendo,i can look at vst imput levels but those meters are not very accurate.
i would think you could do the same in logic but im not familiar with that app.

EDIT: actually i had the vst imputs confused with the vst outputs in nuendo so there is no way in nuendo that i can see how i would accurately tell what the true DAWS imput levels are except by using a track plug in.
unless someone can point out another option.

There really aren't any trim pots on the FF800, unless you're referring to the gain pots on the front four (ok, five) channels...and those are only applicable when using the preamps in the FF. That's obviously very straight forward, however that doesn't apply for my other preamps.

JackAttack - I'm familiar with the format and again, understand it conceptually, it's just making the practical connections (again, no pun intended). However your comment about routing recorded tracks back through the FF is a great idea, something that I never thought of. I appreciate the tip...maybe I can make some progress today.

You guys rock.

sshack, post: 347646 wrote: There really aren't any trim pots on the FF800, unless you're referring to the gain pots on the front four (ok, five) channels...and those are only applicable when using the preamps in the FF. That's obviously very straight forward, however that doesn't apply for my other preamps.

it is straightforward and it does apply to your other preamps wich also have a imput gain knob and for those, maybe also an output level knob.

unless im totally misreading these posts,..you want to visually verify that your recording levels comming into your daws (whether it's from an external pre,onboard pre or a plugged keyboard in the line level imputs or drum mics) are at -18dbfs or whatever you want them to be.
to do that, crank the onboard gain pots(or output levels of amps or synths or?) till you see -18 on the plug.
it is straight forward unless im not understanding your question about recording levels.
that's what i do with my onyx 1200f and my zed r16.
if im way off base here,tell me and put me out of my misery,lol

I control all of my recorded volume from the preamp itself.

If I want a hotter signal I turn it up at the True 8. If I wanted the preamp to run wide open for any reason I would engage the pad on my mixing desk (if I ever unpacked it anymore) or for my mobile rig, I plug a 10db (or 20db or 30db) XLR pad between the mic itself and the mic cable. Actually on most of my microphones there are switchable pads built in but I have plenty without that feature too.

All settings in the DAW are at 0dB. All settings on my mixing desk are at Unity Gain. This gives the cleanest and best possible path for the recorded signal.

As to learning the FF800 itself, running recorded tracks back through it and playing with the routing was the best thing for me personally. It allowed me to bridge the gap between theoretical and practical. I was too used to wires and patchbays for my own good, and I'm by no means the most analog entrenched engineer here.

I do still test drive new routing setups prior to using them at gigs though. I sketch it on a whiteboard or a small notebook, then set it up in my house to check the routing and gear function. If it works it gets a saved preset either on the FF800 and/or on the recording laptop and a small sketch of the settings in my gig notebook. Only then does that setup begin getting used live on gigs. Yeah, it's a little overkill, but I'm still a Gyrene at heart and we don't do anything half way.

I've spent hours today trying different things, still no 'real' luck yet.
However, it's a bit of a different can of worms at this point. I think I'm good on the gain staging at the moment, but now I'm trying to incorporate the outboard gear, just reverb for now, and the routing in/through the FF is really tough for me to figure out.
So in the most basic sense, I'm trying to do one of two things:

1) Track WITH reverb.

...and/or...

2) Record a dry track and apply reverb in mix down.

The good news is that in both cases, I can hear the reverb. The bad news is, once I bounce the track down to an .mp3, it's completely dry. I did manage to bounce a track once and successfully apply the reverb, but it was fully engulfed; I'm not sure how that happened. So the challenge now is to accurately apply and/or monitor/mix a track with outboard devices. I have a compressor coming too...awesome that I'm stuck. :-)

I may just have to call RME tech support on this. *whew*

Definitely apply reverb only in the mixing stage.

Reverb is extremely sensitive to changes in the mix... sometimes it gets buried in other musical elements, or often gets exposed as too strong during quieter spots. Impossible to know which during the tracking phase. And really, what purpose is served by committing reverb to print during tracking? None!
Also, I have found that the process of dithering can often alter the perception of reverb in a mix (I don't pretend to know why), usually exaggerating it. So you need to listen to your FINAL mixes and adjust reverb accordingly.

Jeff

This is definitely where live mix down helps. You can't really record the reverb to a separate track. The signal just isn't strong enough and it never sounds the same on playback. You need to be able to listen to and record the mixdown as it happens.

Just got my FF800, and yet to get it it fully mobilized....
but I've been studying the manual constantly.

The way I understand it is that the totalmix works as a HP/Mon router as described, but can also allow for routing signals to analog outs (and back in) for outbaord effects. In your case, boxcar, I would record the track dry (and maybe send the vocal out to the reverb for the singer), and re-send that dry recorded track to the 'verb for re-recording. Now you have a dry and a verb track to blend as needed.

Like Jack said, you'll need to plan the routing and necessary cabling ahead of time.
I like to have multiple sketches/plans long before I do the project for this exact reason. Usually they get ripped up.
Will the bass, kick, snare, or vocalist need some compression?

Play w/ all of the above. Record dry, send to outboard verb and re-record. Try sending a 'verb'd signal to the vocalist. Try just playing w/ the verb w/in a mix.
Finally, try this: set up an aux/send/fx channel in the DAW. Put your plug reverb on that. Send the vocal to that channel, and bring the fader up as needed. Kind of like the original vox blended w/ the recorded verb vox, but different.
Everything's different. You just gotta play.

thatjeffguy, post: 347709 wrote: Definitely apply reverb only in the mixing stage.

Reverb is extremely sensitive to changes in the mix... sometimes it gets buried in other musical elements, or often gets exposed as too strong during quieter spots. Impossible to know which during the tracking phase. And really, what purpose is served by committing reverb to print during tracking? None!
Also, I have found that the process of dithering can often alter the perception of reverb in a mix (I don't pretend to know why), usually exaggerating it. So you need to listen to your FINAL mixes and adjust reverb accordingly.

Jeff

I agree, I just wanted to see what I could pull off. Ultimately, I'll probably always track dry and just apply effects during mixing. It seems easy enough to put something like reverb on a track for a singer, etc.

Soap, per your quote about recording dry, send to outboard and re-record...logistically I'm not following how to do that. However, it sounds EXACTLY like what I need to do. Would you mind taking another stab at explaining that please?

Thanks all.

sending and re-recording with an interface, goes like this.
this works better for compressors but also works for reverbs in a pinch.
send your recorded dry track through an analog output on the FF to the imput of the compressor or reverb.
take the output of the compressor and send that back to a channel imput on the FF and record that imput to another track.
pull the slider down on the dry track and monitor the new recording track.

you can also do it through digital in/out if you outboard gear has it.

boxcar, post: 347740 wrote: pull the slider down on the dry track and monitor the new recording track.

You'll want to make sure it's a pre fader send. Otherwise the reverb will drop when you pull the dry track fader down.

There's one thing that ruins this whole concept for me. Once you have the reverb on tape, you can't fine tune it. You lost a lot of control over the texture of the wet/dry mix.

yeah, that's not a great way to record reverb because you lose the wet dry mix. re-recording direct like that.(serial recording).then your stuck with it.
that works pretty good for compressors and e.q's and such.
you need(parallel recording) for reverb so your blending it in with the dry track.
i mix OTB so my reverbs are on the mixer sends and i can blend them into the mix.
before i mixed OTB, i used a lexicon mx 300,it had usb and would work as a plug in in my app.so i could put it on the plug in sends.

I would leave the dry fader where it's at, and just mute it in your monitor mix.
Only reason to do it this way is if you prefer your outboard verb to any plugins.
When you brought it up, I started imagining how I would do it, if I could.

Compression is a whole different story. I prefer to print a little on the really transient/punchy sources, and do the rest in the box.
Then again, my dbx's are great for what they do, but necessarily worthy of a send and re-record.

sshack, post: 347132 wrote:
For the most part, I rarely track with any EQ, comp, or limiting, save using my LA-610, but even then it's minimal. My preference is to record as 'organically' as possible (no, I'm not a hippie) and to allow some room for compression, etc. later in the mixing stage. I've really searched for guidance on this and just haven't been able to get a solid answer. It seems that a lot of tutorials that I see tend to revolve around software instruments of which I don't use other than the occasional keyboard plugin. That said, all of my recordings are for acoustic/electric guitar, vox, bass and drums.

Hey - I am a real old-school engineer with decades of analog experience. Now I use ProTools on a Digi002. When it comes to gain staging the main thing to remember is using 0 dB as your reference point. but let me start at the top.

I don't know much about Logic or other DAWS, but I know in ProTools the faders only affect playback volume, not recording levels. Now, old recording mixers actually did use the faders for level control, so Logic could be different from ProTools. but I doubt it.

But anyway - let's look at mics and especially drums.

Mics are either dynamic or condensor (or ribbon, rare). A dynamic mic has an "organic" sound level of about -45 dB by itself at the diaphragm, powered condensor mics are close to -35 dB (note dB is both an acoustic and an electronic measurement - we are speaking of electronic at the moment. Since most audio equipment uses 0 dB as a working reference point, that means your goal with a preamp is to raise the output of your mic to 0dB (ideally) - in the tape days this was critical as we were always fighting tape hiss. Now it is far less critical with digital recording - but the same concept of gain staging applies.

What you want to avoid is distortion from peaks - especially in drums. So, with drums we would generally track at abourt -30 to -20 dB (on a VU Meter, which we used to use all the time on every track). That was because we knew a snare drum would have peaks of about 25 dB that were too fast for a VU meter to register. These days it isn't tape staurtation we worry about so much, but you do have to worry about electronic clipping. So, it is better to track somewhat lower in digital, say -10 dB as a rule, because clipping is more of concern than tape hiss. You also have to think about electronic hiss from outboard gear.

Now - in Protools all tracking is done from the mic >> preamp >> to recording media (not through the fader), so what you see on the meters is the recorded volume despite what the faders may say. That is what you WANT to see on individual tracks - the recorded level. The faders come into play when you are doing a mix and you are looking at the stereo masters. If you had a VU meter to read what your track was doing with the gain from the EQ and compression you would be turning down the fader in most cases because outboard gear usually adds gain. You do want to re-shoot for about 0 dB on your combined mix levels because your mix should be the same volume as every one else's music.

Getting back to recording - keep in mind that if you track too low, that if your mic is running through a preamp that you may find some electronic hiss on your track when you go to add EQ and compression. It could become a real problem, because a pre-amp is going to add noise when you boost he volume of your track if you did not boost the signal from the mic enough. So, as always, the goal is to track as loud as possible before you hit clipping. That gives you more freedom to use EQ and compression withouit the electronic hiss ever being a factor.

Gain stages : Mic (-35 dB) > preamp (raise to -10 to 0 dB) >> recording media. In the Mix; add EQ and compression (this can add 10 to 20 dB of gain) and then adjust the fader level down so the recorded track fits in the mix, with all combined tracks together, adjust your meters to read close to 0 dB. That was how we always did it.

I agree generally except the aiming for 0dB while tracking. In the digital world this does cause problems with headroom as more and more tracks are added into the mix. By the time your mix goes to a Mastering Engineer there should still be enough headroom for the ME to work his/her magic. If the mix is already at 0dB then there is nowhere to go.

I once remember in my electronics class way back in college that in gain staging, the total gain is added if each of those amplifiers or equipments in sequence are given in terms of decibels. But if the gain is given in terms of plain ratio, it will multiplied together to get a total gain. For example, if you have series of amplifiers with gain 2dB, 3dB, 1dB , the total gain is 2+3+1 = 6dB however if its not in dB. Like gain of 4, 3, 2, the total gain is 4x3x2= 24. To convert in dB is to take the logarithm of the values. This can be useful if you have use amplifiers in sequence.

So to avoid clipping in the receiving end, say your input signal is -6dB and if you have +6dB total gain , the final sound output level is 0. Obviously you should give some headroom particularly if you are doing mixing or audio mastering work.

Well, I think I have the gain staging worked out fairly well with keeping my faders at "0", but I'm having a helluva time incorporating the outboard gear. I thought I had it, but ended up being wrong. Almost positive that there's something in Total Mix that I'm missing or doing wrong. I'll just try and give RME a call tomorrow to see if they can walk me through the proper set up, etc.

Thank you to everyone for the insight and discussion around this topic, it's been quite helpful.

The FF series is not set up with inserts at all. So to use outboard gear you would route signal from your line or mic input directly to an analog output one for one. That signal goes through your fx whatever and back in a pair of line inputs. It really isn't designed for live performance for sure.

So in the matrix view let's say you are recording a stereo keyboard (whatever-it'll work the same).

The physical routing looks like this:
Keyboard-->FF800 line 1/line 2 inputs via TRS
FF800 line1 out/line2 out TRS-->FX unit
FX unit-->FF800 line 3 input/line 4 input via TRS

The Totalmix routing via matrix view:
On the top left is AN1. Check the box corresponding to the top row AN1
Left side AN2. Check the box corresponding to the top row AN2

In your DAW:
To record both dry tracks and wet tracks you will need four tracks.
Track1-input from Analog 1 (keyboard left)
Track2-input from Analog 2 (keyboard right)
Track3-input from Analog 3 (wet left)
Track4-input from Analog 4 (wet right)

JackAttack, thanks. That's exactly how I've been cabled up (it's not THAT hard), but my issue was in Total Mix. Some how or another I had a few different tracks associated to others, by accident perhaps and they were screwing up the routing. I basically cleared all of my setting and it worked.

Now I can actually use my compressor and hear what it sounds like. Rock!

JackAttack, thanks. That's exactly how I've been cabled up (it's not THAT hard), but my issue was in Total Mix. Some how or another I had a few different tracks associated to others, by accident perhaps and they were screwing up the routing. I basically cleared all of my setting and it worked.

Now I can actually use my compressor and hear what it sounds like. Rock!

My professor taught me a song about signal routing, I dont think you'll get the melody, but its basically a preschool song for recording. Its called, "Outputs to Inputs".

Erm em em em...
1.. 2... 1, 2, 3 "Outputs to inputs, outputs to inputs (thats the main jingle). You go out from the microphone, in to the snake box. Out of the snake box and in to the patch box (bay). Out of the patch box and in to the pre amps. Out of the pre amps and in to the converter. Out of the converter and in to the computer. And that's when you turn it all over. Out of the computer and in to the converter, out of the converter and into the amp..."

This song is terrible, he made us sing it anytime we got something screwed up. The console song was even worse. A song about a channel strip from top to bottom...

That's an interesting song Todd, however it really has nothing to do with the original post nor does it really have anything to do with gain staging. I think this thread is a little more complicated than the header reveals.

TheJackAttack, post: 347898 wrote: I agree generally except the aiming for 0dB while tracking. In the digital world this does cause problems with headroom as more and more tracks are added into the mix. By the time your mix goes to a Mastering Engineer there should still be enough headroom for the ME to work his/her magic. If the mix is already at 0dB then there is nowhere to go.

I bet he means 0dbu, which could translate to anything from -24dBFS to -14dBFS. On the setup I usually use 0dBu on the analog meter is about -18dBFS.