Hardware insert clipping on Mix Bus

OK, I did it. As I mixed in Cubase I screwed up my gain staging so that I occasionally clip the output of the mix bus. No problem right? Just pull down the fader.

However, I use a hardware EQ and a hardware compressor as inserts on my mix bus. So, since they are pre fader they keep getting clipped.

How do you deal with this? Do you just insert something like a Waves L1 before the hardware? Find the offending transient? Start over?

BTW: I can’t hear the clipping as it goes out of my Aurora 16 and into my monitors. There is no digital distortion sound. So, I’m not sure if this is even an issue. Still, I was wondering how people are dealing with this kind of situation,

Thanks for your replies and insights.

You could create one more group routed to the 2-buss; route all tracks to it instead of the 2-buss; then pull down the new group’s fader so the signal to the mix bus (and its inserts) isn’t as hot.

Alternatively, you could turn down the input gain at the top of the mix bus (I think it has one?).

Thanks Alexis. I’ll check into doing both.

BTW: I have always wondered how manipulating the gain within the Cubase mixer changes (or doesn’t change) the original input and recorded sound(s). Is there any science explaining what is going on? If you know, I’d appreciate your-or anybody else’s- thoughts.

The high science behind it is: negative gain values lower volume - positive values raise volume.

There’s quite a nice signal flow diagram in the manual; ‘VST Mixer Diagrams’.

Thanks for clarifying. I got spooked a long time ago about levels from this thread:

http://www.gearslutz.com/board/so-much-gear-so-little-time/463010-reason-most-itb-mixes-don-t-sound-good-analog-mixes-restored.html

Hence, my “science” question. There’s a lot of talk about voltages and a statement that says, “Running a Digital mix right to the top of the scale is like running your SSL mix buss where the VU meters are slammed all the way to the right and you are constantly hitting it at +25. No one will get a good sounding running the desk like that. You won’t get a good sounding mix in digital either.”

So, I ask .

The funny thing is, even with my Aurora 16 showing the occasional transient clip, I cannot hear any distortion.

That was all based on a fixed integer mix bus (old pro tools), almost all DAW’s these days use floating point mix buses and does away with the worry of any sort of clipping in the digital domane.

None the less, to be correct you shouldn’t be going over 0dBFS (digital) when going back to the analog domane even if you cannot hear the clipping.

Forget that thread. It includes more misinformation than all other GearSnobs threads combined. And that’s A LOT. :smiling_imp:

Jarno: Could you elaborate on some of the misinformation? There are still disciples saying that there must be an ideal
RMS/peak input to create I suppose some kind of perfect waveform for mixing. I don’t record hot or low and will use my trim if I have any problems like the one in my original post, but they’re holding onto this ideal window and claiming if you do it your mix will sound better.

OK. But just from the first page. Don’t want to go through 150+ pages … again.

Running a Digital mix right to the top of the scale is like running your SSL mix buss where the VU meters are slammed all the way to the right and you are constantly hitting it at +25.

Complete rubbish. Every DAW works just fine with signals hitting up to 0dBfs. And modern floating-point DAWs (and well designed 32+ bit fixed-point mix engines too) can go further.

Plugins use the same reference at real equipment. Never try and drive them to the top of the Digital scale.

No they don’t. Most modern plugins have linear response in 10s or 100s of dBs of operating range. Only very few ones have “sweet spot”.

Set your AD convertors to 0 dBvu = minus 20 dBfs and record your tracks at an average level of -20 dBfs and everything will fall into place.

And by doing this you may throw away a significant part of your dynamic range or even clip your analog equipment/converters … depending on the audio material, of course.

Your DAW doesn’t care. There are some plugins with “sweet spots”, but if you’re using them, you should know what to do.

And there are still disciples saying that digital audio signal consists of “steps” and to smooth out those “steps” you need at least 192k sampling frequency. :unamused:

Thanks again for your reply Jarno.

I did learn one thing that I didn’t know. My UAD plugins have an “internal operating level”. So, for anybody reading this and using UAD plugs, here they are.

And yes, since some plugs work at -12 dBFS and others -18dBFS, if you are using them in succession you have to change the output of the first plug so that it meets the internal operating level of the next plug.


Yes. All those plugins listed on that table are emulations of analog gear and are programmed to have a “sweet spot”. But remember: When their “operating level” is -18dBfs, they have “headroom” of 18dB, which means they are capaple of processing signals peaking at 0dBfs.

Not necessarily. It all depends on how hard you want to “drive” the plugin. Only real rule is: don’t exceed 0dBfs, if your plugin isn’t supporting levels above it … the rest is only matter of taste (your ears).

Thanks Jarno. I’ve never even considered a “sweet spot” for my UAD 1176. So, I’ll check it out. Though I never paid any attention to levels, except making sure not to clip my Cubase channels, groups and busses, I use the 1176 to crush my drum room mics and it sounds great blended with the rest of the kit. It will be interesting to see what this sweet spot sounds like. I am skeptical, but we’ll see. Perhaps it would be more apparent on a vocal.