Preamp Outs Clipping on my AD Converter ins, but not in DAW. Is it bad?

Question

TL;DR: Do I need to avoid clipping on my AD Converter as well as the input in my DAW when gain staging? how much headroom should I leave on a track once it leaves my Preamp, and hits my AD Converter ins? Even though it was clipping on my AD Converters, it sounded full and warm to me? What's the point of having good sounding pre's that you can't run a little hot? Is the +20 dBu max limit on my converters basically the line in the sand that says "sorry, turn your pre's down, the ADC's can't handle it." ? Ahh so many questions!! (Rhode Island is neither a road, nor an Island.. discuss!)

Hi friends, Lots to unpack here... but ultimately this is another dumb gain staging post! I feel dumb asking this, because I understand its likely a very annoying question, but my google-fu is failing me and I need some guidance. I haven't been able to find much on gain staging when it comes to ADC converters. I have done a fair amount of mixing in the box, mixing someone else's recorded instruments for albums, or mixing line level stuff like drum machines/synths etc. However, this is my first time trying to actually attempting to engineer a recording for my band, and mix the album start to finish. I understand that gain staging is very important, and that you want plenty of headroom to work with wile mixing. One of the main things I don't understand right now: What is a reasonable db gain level when tracking dynamic microphones plugged into a preamp, that then routes directly to your interface inputs? A little details about my setup:

AD/DA Converter & Interface: Orion 32+

Preamps: Neve 1073OPX

Mics: mostly all SM57's, AKG C414 XLII pair, Senn. 421, audix d6, diy subkick.

So here's where my confusion comes in. I mic'd my kit up the other day to prep it for tracking next week. Setting the gain levels on the 1073OPX channels, the gain was set on each channel all roughly around 40-50db, with the -25db pad set on each. The preamps sounded best when I was monitoring them around ~50-60db. Felt like this gave the drums the most color/"warmth", but when I looked at my Antelope interface software, many of the db meters were hitting the red on the line ins. When those line ins were sent to an ASIO input to cubase, those inputs were not clipping (I assume this is something to do with 32/64bit float) . When I zoom in on those tracks, I could see the tops of the waveforms were lopped off, obviously at the ADC. I can definitely track all of my channels with way less gain makeup, but is this the right approach? Or am I missing something in between the pre's and the interface to attenuate the signal before it hits the interface?

Some specs on my interface that stood out to me while I was looking into this:

Analog Inputs: 4 x D-SUB 25 (32 channels total), +20 dBu max**,** 11.2 kOhms

A/D Converter Dynamic Range: 118 dB THD + N: -105 dB

Is this +20 dBu max basically the line in the sand that says "sorry, turn your Pre's down, the ADC's can't handle it." ? I do plan on printing the recorded tracks through some outboard compressors etc.

I realize this post is kind of all over the place. Just looking for some guidance from the pro's. Love you all! Thanks for taking the time to read :D

Answer 1

This is really a fundamental question around recording technique for bands, of which there are a number of schools of thought that are equally applicable.

Personally, I come from a classical music background where no audio processing is applied to the signal at all before conversion, however band recording is often different and the usually accepted technique is to use input compression prior to conversion.

Using input compression prior to conversion can help tame transients and avoid situations where ADC clipping can occur.

If you are not using compression (vocal tracks, drum tracks etc.) then you must be careful to record everything within the dynamic range of the converter.

Each microphone you use will have a different output signal level and you will need to tweak to taste based on mic positioning, angle and peak sound pressure level being presented at the capsule. There really isn't a generic formula, other than to treat 0dBU as 0VU on a traditional analogue meter as a gain staging starting point.

This will give you 20dB of headroom to play with. As you have noticed, above 20dBU is a no-go with digital audio - you will basically get clipped and lose information. With analogue, you have less headroom to play with and have to deal with tape distortion and other nasties, which in some situations can be a creative tool. You don't have these options in the digital world.

My best recommendation is to start at "+30dB" on the mic input preamp, tweak up or down to taste, don't peak above -6dBFS on the ADC and go from there. I would also consider applying compression to particularly "peaky" tracks prior to conversion - this may or may not be an option for you. This will also depend a lot on the genre of music you are recording and how 'compressed' this is going to end up sounding in the mix. In these cases, a little compression added at the front-end can give you more options in the mix and a better overall dynamic range during conversion.

HTH

Answer 2

After much research, I opted to put a passive attenuator (Little Labs redcloud) in between my preamp and interface. This allows me to run the preamps hot to get additional saturation without clipping at the ADC. The LL redcloud let’s me drop the output level of the preamp without coloring the signal at all. All of your input has been much appreciated!

Answer 3

Given that you are using the 1073 mic pre-amps (the best pre-amp in the world, and you aren't even using a clone, the actual Neve version) if your pre-amp tells you, you are clipping, I would listen to it. Especially a 1073, which is the best pre-amp in the world, everyone from Kanye West to Alanis Morrissette used a 1073 to record. And when you say your DAW uses 32-bit and 64-bit, that is half wrong, your DAW uses a 32-bit floating point wavform encoding method, which is gives the exact same bit-depth as 24-bit, only it allows an extra 8-bits above the 24bit level to prevent digital clipping (which is why I'm guessing your DAW isn't clipping, but your pre-amps are), where as working in 24 bit depth strictly in your DAW means anything over 0dbFS means it will digitally distort(completely and entirely undesirable), with most modern DAWs working in 32-bit depth that allows an extra 8-bits over the 24bits to prevent digital clipping, which I'm pretty sure is happening in your situation. But IGNORE the digital end of the equation, you are using THE ABSOLUTE BEST pre-amps in the world, and if they are signaling that you are clipping then listen to them. In fact I'm not even sure how you got those pre-amps to clip, as I will ONLY USE 1073's to pre-amplify a microphone's signal, and they have NEVER clipped on me, I have even driven a super hot signal into them to get the benefit of the coloration that the 1073's will provide to a signal. And Neve has built them so that driving a super hot signal into them is almost a given, that's why they have an input gain control and an output gain control. So, like I just stated, I'm not sure how you got the 1073's to clip, as mine as yet to clip(although I AM an educated Audio Engineer, and Electrical Engineer so I would never set up my mic's improperly), maybe something's wrong in your chain into your DAW, can you pls detail your chain into your DAW.